Merge "Remove test mode from VirtualCameraRenderThread." into main
diff --git a/camera/camera_platform.aconfig b/camera/camera_platform.aconfig
index ab161d6..fe10e12 100644
--- a/camera/camera_platform.aconfig
+++ b/camera/camera_platform.aconfig
@@ -237,3 +237,13 @@
purpose: PURPOSE_BUGFIX
}
}
+
+flag {
+ namespace: "camera_platform"
+ name: "multi_res_raw_reprocessing"
+ description: "Allow multi-resolution raw reprocessing without reprocessing capability"
+ bug: "336922859"
+ metadata {
+ purpose: PURPOSE_BUGFIX
+ }
+}
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 28670b1..03c765a 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -362,8 +362,8 @@
const ui::DisplayState& displayState,
const sp<IGraphicBufferProducer>& bufferProducer,
sp<IBinder>* pDisplayHandle, sp<SurfaceControl>* mirrorRoot) {
- sp<IBinder> dpy = SurfaceComposerClient::createDisplay(
- String8("ScreenRecorder"), gSecureDisplay);
+ static const std::string kDisplayName("ScreenRecorder");
+ sp<IBinder> dpy = SurfaceComposerClient::createVirtualDisplay(kDisplayName, gSecureDisplay);
SurfaceComposerClient::Transaction t;
t.setDisplaySurface(dpy, bufferProducer);
setDisplayProjection(t, dpy, displayState);
@@ -797,7 +797,7 @@
sp<Overlay> overlay;
~RecordingData() {
- if (dpy != nullptr) SurfaceComposerClient::destroyDisplay(dpy);
+ if (dpy != nullptr) SurfaceComposerClient::destroyVirtualDisplay(dpy);
if (overlay != nullptr) overlay->stop();
if (encoder != nullptr) {
encoder->stop();
diff --git a/media/codec2/hal/plugin/FilterWrapper.cpp b/media/codec2/hal/plugin/FilterWrapper.cpp
index 197d6e7..b926150 100644
--- a/media/codec2/hal/plugin/FilterWrapper.cpp
+++ b/media/codec2/hal/plugin/FilterWrapper.cpp
@@ -49,11 +49,6 @@
std::weak_ptr<FilterWrapper> filterWrapper)
: mIntf(intf), mFilterWrapper(filterWrapper) {
takeFilters(std::move(filters));
- for (size_t i = 0; i < mFilters.size(); ++i) {
- mControlParamTypes.insert(
- mFilters[i].desc.controlParams.begin(),
- mFilters[i].desc.controlParams.end());
- }
}
~WrappedDecoderInterface() override = default;
@@ -91,6 +86,12 @@
// TODO: documentation
mFilters = std::move(filters);
+ mControlParamTypes.clear();
+ for (size_t i = 0; i < mFilters.size(); ++i) {
+ mControlParamTypes.insert(
+ mFilters[i].desc.controlParams.begin(),
+ mFilters[i].desc.controlParams.end());
+ }
mTypeToIndexForQuery.clear();
mTypeToIndexForConfig.clear();
for (size_t i = 0; i < mFilters.size(); ++i) {
diff --git a/media/codec2/sfplugin/CCodec.cpp b/media/codec2/sfplugin/CCodec.cpp
index 20b6d7f..a897fa0 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -2229,9 +2229,15 @@
// See also b/300350761.
//
// The workaround is no longer needed with fetchGraphicBlock & C2Fence changes.
- // so we are reverting back to the logical sequence of the operations.
+ // so we are reverting back to the logical sequence of the operations when
+ // AIDL HALs are selected.
+ // When the HIDL HALs are selected, we retained workaround(the reversed
+ // order) as default in order to keep legacy behavior.
+ bool stopHalBeforeSurface =
+ Codec2Client::IsAidlSelected() ||
+ property_get_bool("debug.codec2.stop_hal_before_surface", false);
status_t err = C2_OK;
- if (android::media::codec::provider_->stop_hal_before_surface()) {
+ if (stopHalBeforeSurface && android::media::codec::provider_->stop_hal_before_surface()) {
err = comp->stop();
mChannel->stopUseOutputSurface(pushBlankBuffer);
} else {
@@ -2334,8 +2340,14 @@
// See also b/300350761.
//
// The workaround is no longer needed with fetchGraphicBlock & C2Fence changes.
- // so we are reverting back to the logical sequence of the operations.
- if (android::media::codec::provider_->stop_hal_before_surface()) {
+ // so we are reverting back to the logical sequence of the operations when
+ // AIDL HALs are selected.
+ // When the HIDL HALs are selected, we retained workaround(the reversed
+ // order) as default in order to keep legacy behavior.
+ bool stopHalBeforeSurface =
+ Codec2Client::IsAidlSelected() ||
+ property_get_bool("debug.codec2.stop_hal_before_surface", false);
+ if (stopHalBeforeSurface && android::media::codec::provider_->stop_hal_before_surface()) {
comp->release();
mChannel->stopUseOutputSurface(pushBlankBuffer);
} else {
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 3984b83..c7ab82f 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -2784,7 +2784,16 @@
}
void CCodecBufferChannel::setInfoBuffer(const std::shared_ptr<C2InfoBuffer> &buffer) {
- mInfoBuffers.push_back(buffer);
+ if (mInputSurface == nullptr) {
+ mInfoBuffers.push_back(buffer);
+ } else {
+ std::list<std::unique_ptr<C2Work>> items;
+ std::unique_ptr<C2Work> work(new C2Work);
+ work->input.infoBuffers.emplace_back(*buffer);
+ work->worklets.emplace_back(new C2Worklet);
+ items.push_back(std::move(work));
+ c2_status_t err = mComponent->queue(&items);
+ }
}
status_t toStatusT(c2_status_t c2s, c2_operation_t c2op) {
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 04a8a45..e667964 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -315,6 +315,7 @@
"aidl/android/media/DeviceConnectedState.aidl",
"aidl/android/media/EffectDescriptor.aidl",
"aidl/android/media/SurroundSoundConfig.aidl",
+ "aidl/android/media/TrackInternalMuteInfo.aidl",
"aidl/android/media/TrackSecondaryOutputInfo.aidl",
],
defaults: [
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index aa5c840..cf3b43a 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -918,6 +918,11 @@
return OK;
}
+status_t AudioFlingerClientAdapter::setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& tracksInternalMuted) {
+ return statusTFromBinderStatus(mDelegate->setTracksInternalMute(tracksInternalMuted));
+}
+
////////////////////////////////////////////////////////////////////////////////////////////////////
// AudioFlingerServerAdapter
AudioFlingerServerAdapter::AudioFlingerServerAdapter(
@@ -1477,4 +1482,9 @@
return Status::ok();
}
+Status AudioFlingerServerAdapter::setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& tracksInternalMute) {
+ return Status::fromStatusT(mDelegate->setTracksInternalMute(tracksInternalMute));
+}
+
} // namespace android
diff --git a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
index 31d3af5..e8fcf77 100644
--- a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
@@ -41,6 +41,7 @@
import android.media.ISoundDoseCallback;
import android.media.MicrophoneInfoFw;
import android.media.RenderPosition;
+import android.media.TrackInternalMuteInfo;
import android.media.TrackSecondaryOutputInfo;
import android.media.audio.common.AudioChannelLayout;
import android.media.audio.common.AudioFormatDescription;
@@ -293,6 +294,11 @@
*/
AudioPortFw getAudioMixPort(in AudioPortFw devicePort, in AudioPortFw mixPort);
+ /**
+ * Set internal mute for a list of tracks.
+ */
+ void setTracksInternalMute(in TrackInternalMuteInfo[] tracksInternalMute);
+
// When adding a new method, please review and update
// IAudioFlinger.h AudioFlingerServerAdapter::Delegate::TransactionCode
// AudioFlinger.cpp AudioFlinger::onTransactWrapper()
diff --git a/media/libaudioclient/aidl/android/media/TrackInternalMuteInfo.aidl b/media/libaudioclient/aidl/android/media/TrackInternalMuteInfo.aidl
new file mode 100644
index 0000000..05b1fa4
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/TrackInternalMuteInfo.aidl
@@ -0,0 +1,23 @@
+/*
+ * Copyright (C) 2024 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+parcelable TrackInternalMuteInfo {
+ /* Interpreted as audio_port_handle_t. */
+ int portId;
+ boolean muted;
+}
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 5a1e037..860a0bc 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -60,6 +60,7 @@
#include "android/media/OpenInputResponse.h"
#include "android/media/OpenOutputRequest.h"
#include "android/media/OpenOutputResponse.h"
+#include "android/media/TrackInternalMuteInfo.h"
#include "android/media/TrackSecondaryOutputInfo.h"
namespace android {
@@ -388,6 +389,9 @@
virtual status_t getAudioMixPort(const struct audio_port_v7 *devicePort,
struct audio_port_v7 *mixPort) const = 0;
+
+ virtual status_t setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& tracksInternalMute) = 0;
};
/**
@@ -504,6 +508,8 @@
status_t getAudioPolicyConfig(media::AudioPolicyConfig* output) override;
status_t getAudioMixPort(const struct audio_port_v7 *devicePort,
struct audio_port_v7 *mixPort) const override;
+ status_t setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& tracksInternalMute) override;
private:
const sp<media::IAudioFlingerService> mDelegate;
@@ -606,6 +612,7 @@
GET_AUDIO_POLICY_CONFIG =
media::BnAudioFlingerService::TRANSACTION_getAudioPolicyConfig,
GET_AUDIO_MIX_PORT = media::BnAudioFlingerService::TRANSACTION_getAudioMixPort,
+ SET_TRACKS_INTERNAL_MUTE = media::BnAudioFlingerService::TRANSACTION_setTracksInternalMute,
};
protected:
@@ -742,6 +749,8 @@
Status getAudioMixPort(const media::AudioPortFw& devicePort,
const media::AudioPortFw& mixPort,
media::AudioPortFw* _aidl_return) override;
+ Status setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& tracksInternalMute) override;
private:
const sp<AudioFlingerServerAdapter::Delegate> mDelegate;
};
diff --git a/media/libaudiohal/impl/StreamHalAidl.cpp b/media/libaudiohal/impl/StreamHalAidl.cpp
index 5aff5f2..84c1a8a 100644
--- a/media/libaudiohal/impl/StreamHalAidl.cpp
+++ b/media/libaudiohal/impl/StreamHalAidl.cpp
@@ -285,11 +285,12 @@
return OK;
}
-status_t StreamHalAidl::getObservablePosition(int64_t *frames, int64_t *timestamp) {
+status_t StreamHalAidl::getObservablePosition(int64_t* frames, int64_t* timestamp,
+ StatePositions* statePositions) {
ALOGV("%p %s::%s", this, getClassName().c_str(), __func__);
if (!mStream) return NO_INIT;
StreamDescriptor::Reply reply;
- RETURN_STATUS_IF_ERROR(updateCountersIfNeeded(&reply));
+ RETURN_STATUS_IF_ERROR(updateCountersIfNeeded(&reply, statePositions));
*frames = std::max<int64_t>(0, reply.observable.frames);
*timestamp = std::max<int64_t>(0, reply.observable.timeNs);
return OK;
@@ -446,8 +447,12 @@
if (auto state = getState(); state == StreamDescriptor::State::DRAINING) {
// Retrieve the current state together with position counters unconditionally
// to ensure that the state on our side gets updated.
- sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(),
- nullptr, true /*safeFromNonWorkerThread */);
+ sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(), nullptr,
+ true /*safeFromNonWorkerThread */);
+ // For compatibility with HIDL behavior, apply a "soft" position reset
+ // after receiving the "drain ready" callback.
+ std::lock_guard l(mLock);
+ mStatePositions.framesAtFlushOrDrain = mLastReply.observable.frames;
} else {
ALOGW("%s: unexpected onDrainReady in the state %s", __func__, toString(state).c_str());
}
@@ -455,15 +460,8 @@
void StreamHalAidl::onAsyncError() {
std::lock_guard l(mLock);
- if (mLastReply.state == StreamDescriptor::State::IDLE ||
- mLastReply.state == StreamDescriptor::State::DRAINING ||
- mLastReply.state == StreamDescriptor::State::TRANSFERRING) {
- mLastReply.state = StreamDescriptor::State::ERROR;
- ALOGW("%s: onError received", __func__);
- } else {
- ALOGW("%s: unexpected onError in the state %s", __func__,
- toString(mLastReply.state).c_str());
- }
+ ALOGW("%s: received in the state %s", __func__, toString(mLastReply.state).c_str());
+ mLastReply.state = StreamDescriptor::State::ERROR;
}
status_t StreamHalAidl::createMmapBuffer(int32_t minSizeFrames __unused,
@@ -514,9 +512,9 @@
}
status_t StreamHalAidl::sendCommand(
- const ::aidl::android::hardware::audio::core::StreamDescriptor::Command &command,
+ const ::aidl::android::hardware::audio::core::StreamDescriptor::Command& command,
::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply,
- bool safeFromNonWorkerThread) {
+ bool safeFromNonWorkerThread, StatePositions* statePositions) {
// TIME_CHECK(); // TODO(b/243839867) reenable only when optimized.
if (!safeFromNonWorkerThread) {
const pid_t workerTid = mWorkerTid.load(std::memory_order_acquire);
@@ -548,6 +546,23 @@
}
mLastReply = *reply;
mLastReplyExpirationNs = uptimeNanos() + mLastReplyLifeTimeNs;
+ if (!mIsInput && reply->status == STATUS_OK) {
+ if (command.getTag() == StreamDescriptor::Command::standby &&
+ reply->state == StreamDescriptor::State::STANDBY) {
+ mStatePositions.framesAtStandby = reply->observable.frames;
+ } else if (command.getTag() == StreamDescriptor::Command::flush &&
+ reply->state == StreamDescriptor::State::IDLE) {
+ mStatePositions.framesAtFlushOrDrain = reply->observable.frames;
+ } else if (!mContext.isAsynchronous() &&
+ command.getTag() == StreamDescriptor::Command::drain &&
+ (reply->state == StreamDescriptor::State::IDLE ||
+ reply->state == StreamDescriptor::State::DRAINING)) {
+ mStatePositions.framesAtFlushOrDrain = reply->observable.frames;
+ } // for asynchronous drain, the frame count is saved in 'onAsyncDrainReady'
+ }
+ if (statePositions != nullptr) {
+ *statePositions = mStatePositions;
+ }
}
}
switch (reply->status) {
@@ -563,7 +578,8 @@
}
status_t StreamHalAidl::updateCountersIfNeeded(
- ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply) {
+ ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply,
+ StatePositions* statePositions) {
bool doUpdate = false;
{
std::lock_guard l(mLock);
@@ -573,10 +589,13 @@
// Since updates are paced, it is OK to perform them from any thread, they should
// not interfere with I/O operations of the worker.
return sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(),
- reply, true /*safeFromNonWorkerThread */);
+ reply, true /*safeFromNonWorkerThread */, statePositions);
} else if (reply != nullptr) { // provide cached reply
std::lock_guard l(mLock);
*reply = mLastReply;
+ if (statePositions != nullptr) {
+ *statePositions = mStatePositions;
+ }
}
return OK;
}
@@ -668,8 +687,19 @@
return BAD_VALUE;
}
int64_t aidlFrames = 0, aidlTimestamp = 0;
- RETURN_STATUS_IF_ERROR(getObservablePosition(&aidlFrames, &aidlTimestamp));
- *dspFrames = aidlFrames;
+ StatePositions statePositions{};
+ RETURN_STATUS_IF_ERROR(
+ getObservablePosition(&aidlFrames, &aidlTimestamp, &statePositions));
+ // Number of audio frames since the stream has exited standby.
+ // See the table at the start of 'StreamHalInterface' on when it needs to reset.
+ int64_t mostRecentResetPoint;
+ if (!mContext.isAsynchronous() && audio_has_proportional_frames(mConfig.format)) {
+ mostRecentResetPoint = statePositions.framesAtStandby;
+ } else {
+ mostRecentResetPoint =
+ std::max(statePositions.framesAtStandby, statePositions.framesAtFlushOrDrain);
+ }
+ *dspFrames = aidlFrames <= mostRecentResetPoint ? 0 : aidlFrames - mostRecentResetPoint;
return OK;
}
@@ -726,8 +756,16 @@
return BAD_VALUE;
}
int64_t aidlFrames = 0, aidlTimestamp = 0;
- RETURN_STATUS_IF_ERROR(getObservablePosition(&aidlFrames, &aidlTimestamp));
- *frames = aidlFrames;
+ StatePositions statePositions{};
+ RETURN_STATUS_IF_ERROR(getObservablePosition(&aidlFrames, &aidlTimestamp, &statePositions));
+ // See the table at the start of 'StreamHalInterface'.
+ if (!mContext.isAsynchronous() && audio_has_proportional_frames(mConfig.format)) {
+ *frames = aidlFrames;
+ } else {
+ const int64_t mostRecentResetPoint =
+ std::max(statePositions.framesAtStandby, statePositions.framesAtFlushOrDrain);
+ *frames = aidlFrames <= mostRecentResetPoint ? 0 : aidlFrames - mostRecentResetPoint;
+ }
timestamp->tv_sec = aidlTimestamp / NANOS_PER_SECOND;
timestamp->tv_nsec = aidlTimestamp - timestamp->tv_sec * NANOS_PER_SECOND;
return OK;
diff --git a/media/libaudiohal/impl/StreamHalAidl.h b/media/libaudiohal/impl/StreamHalAidl.h
index 8a398d8..fff7a92 100644
--- a/media/libaudiohal/impl/StreamHalAidl.h
+++ b/media/libaudiohal/impl/StreamHalAidl.h
@@ -194,6 +194,11 @@
// For tests.
friend class sp<StreamHalAidl>;
+ struct StatePositions {
+ int64_t framesAtFlushOrDrain;
+ int64_t framesAtStandby;
+ };
+
template<class T>
static std::shared_ptr<::aidl::android::hardware::audio::core::IStreamCommon> getStreamCommon(
const std::shared_ptr<T>& stream);
@@ -212,7 +217,8 @@
status_t getLatency(uint32_t *latency);
// Always returns non-negative values.
- status_t getObservablePosition(int64_t *frames, int64_t *timestamp);
+ status_t getObservablePosition(int64_t* frames, int64_t* timestamp,
+ StatePositions* statePositions = nullptr);
// Always returns non-negative values.
status_t getHardwarePosition(int64_t *frames, int64_t *timestamp);
@@ -268,11 +274,13 @@
// Note: Since `sendCommand` takes mLock while holding mCommandReplyLock, never call
// it with `mLock` being held.
status_t sendCommand(
- const ::aidl::android::hardware::audio::core::StreamDescriptor::Command &command,
+ const ::aidl::android::hardware::audio::core::StreamDescriptor::Command& command,
::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply = nullptr,
- bool safeFromNonWorkerThread = false);
+ bool safeFromNonWorkerThread = false,
+ StatePositions* statePositions = nullptr);
status_t updateCountersIfNeeded(
- ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply = nullptr);
+ ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply = nullptr,
+ StatePositions* statePositions = nullptr);
const std::shared_ptr<::aidl::android::hardware::audio::core::IStreamCommon> mStream;
const std::shared_ptr<::aidl::android::media::audio::IHalAdapterVendorExtension> mVendorExt;
@@ -280,6 +288,9 @@
std::mutex mLock;
::aidl::android::hardware::audio::core::StreamDescriptor::Reply mLastReply GUARDED_BY(mLock);
int64_t mLastReplyExpirationNs GUARDED_BY(mLock) = 0;
+ // Cached values of observable positions when the stream last entered certain state.
+ // Updated for output streams only.
+ StatePositions mStatePositions GUARDED_BY(mLock) = {};
// mStreamPowerLog is used for audio signal power logging.
StreamPowerLog mStreamPowerLog;
std::atomic<pid_t> mWorkerTid = -1;
@@ -328,10 +339,14 @@
// Requests notification when data buffered by the driver/hardware has been played.
status_t drain(bool earlyNotify) override;
- // Notifies to the audio driver to flush the queued data.
+ // Notifies to the audio driver to flush (that is, drop) the queued data. Stream must
+ // already be paused before calling 'flush'.
status_t flush() override;
// Return a recent count of the number of audio frames presented to an external observer.
+ // This excludes frames which have been written but are still in the pipeline. See the
+ // table at the start of the 'StreamOutHalInterface' for the specification of the frame
+ // count behavior w.r.t. 'flush', 'drain' and 'standby' operations.
status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp) override;
// Notifies the HAL layer that the framework considers the current playback as completed.
@@ -413,6 +428,7 @@
// Return a recent count of the number of audio frames received and
// the clock time associated with that frame count.
+ // The count must not reset to zero when a PCM input enters standby.
status_t getCapturePosition(int64_t *frames, int64_t *time) override;
// Get active microphones
diff --git a/media/libaudiohal/impl/StreamHalHidl.h b/media/libaudiohal/impl/StreamHalHidl.h
index 80379d0..433e0a3 100644
--- a/media/libaudiohal/impl/StreamHalHidl.h
+++ b/media/libaudiohal/impl/StreamHalHidl.h
@@ -161,10 +161,14 @@
// Requests notification when data buffered by the driver/hardware has been played.
virtual status_t drain(bool earlyNotify);
- // Notifies to the audio driver to flush the queued data.
+ // Notifies to the audio driver to flush (that is, drop) the queued data. Stream must
+ // already be paused before calling 'flush'.
virtual status_t flush();
// Return a recent count of the number of audio frames presented to an external observer.
+ // This excludes frames which have been written but are still in the pipeline. See the
+ // table at the start of the 'StreamOutHalInterface' for the specification of the frame
+ // count behavior w.r.t. 'flush', 'drain' and 'standby' operations.
virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
// Notifies the HAL layer that the framework considers the current playback as completed.
@@ -259,6 +263,7 @@
// Return a recent count of the number of audio frames received and
// the clock time associated with that frame count.
+ // The count must not reset to zero when a PCM input enters standby.
virtual status_t getCapturePosition(int64_t *frames, int64_t *time);
// Get active microphones
diff --git a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
index eb14f6b..585a895 100644
--- a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
@@ -135,6 +135,38 @@
virtual ~StreamOutHalInterfaceLatencyModeCallback() = default;
};
+/**
+ * On position reporting. There are two methods: 'getRenderPosition' and
+ * 'getPresentationPosition'. The first difference is that they may have a
+ * time offset because "render" position relates to what happens between
+ * ADSP and DAC, while "observable" position is relative to the external
+ * observer. The second difference is that 'getRenderPosition' always
+ * resets on standby (for all types of stream data) according to its
+ * definition. Since the original C definition of 'getRenderPosition' used
+ * 32-bit frame counters, and also because in complex playback chains that
+ * include wireless devices the "observable" position has more practical
+ * meaning, 'getRenderPosition' does not exist in the AIDL HAL interface.
+ * The table below summarizes frame count behavior for 'getPresentationPosition':
+ *
+ * | Mixed | Direct | Direct
+ * | | non-offload | offload
+ * ==============|============|==============|==============
+ * PCM and | Continuous | |
+ * encapsulated | | |
+ * bitstream | | |
+ * --------------|------------| Continuous† |
+ * Bitstream | | | Reset on
+ * encapsulated | | | flush, drain
+ * into PCM | | | and standby
+ * | Not | |
+ * --------------| supported |--------------|
+ * Bitstream | | Reset on |
+ * | | flush, drain |
+ * | | and standby |
+ * | | |
+ *
+ * † - on standby, reset of the frame count happens at the framework level.
+ */
class StreamOutHalInterface : public virtual StreamHalInterface {
public:
// Return the audio hardware driver estimated latency in milliseconds.
@@ -173,10 +205,14 @@
// Requests notification when data buffered by the driver/hardware has been played.
virtual status_t drain(bool earlyNotify) = 0;
- // Notifies to the audio driver to flush the queued data.
+ // Notifies to the audio driver to flush (that is, drop) the queued data. Stream must
+ // already be paused before calling 'flush'.
virtual status_t flush() = 0;
// Return a recent count of the number of audio frames presented to an external observer.
+ // This excludes frames which have been written but are still in the pipeline. See the
+ // table at the start of the 'StreamOutHalInterface' for the specification of the frame
+ // count behavior w.r.t. 'flush', 'drain' and 'standby' operations.
virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp) = 0;
// Notifies the HAL layer that the framework considers the current playback as completed.
@@ -270,6 +306,7 @@
// Return a recent count of the number of audio frames received and
// the clock time associated with that frame count.
+ // The count must not reset to zero when a PCM input enters standby.
virtual status_t getCapturePosition(int64_t *frames, int64_t *time) = 0;
// Get active microphones
diff --git a/media/libmediaplayerservice/fuzzer/Android.bp b/media/libmediaplayerservice/fuzzer/Android.bp
index 7a1411d..0031855 100644
--- a/media/libmediaplayerservice/fuzzer/Android.bp
+++ b/media/libmediaplayerservice/fuzzer/Android.bp
@@ -44,7 +44,7 @@
],
fuzz_config: {
cc: [
- "android-media-fuzzing-reports@google.com",
+ "android-media-playback+bugs@google.com",
],
componentid: 155276,
hotlists: [
diff --git a/media/libstagefright/data/media_codecs_sw.xml b/media/libstagefright/data/media_codecs_sw.xml
index d50bc1e..6fb9232 100644
--- a/media/libstagefright/data/media_codecs_sw.xml
+++ b/media/libstagefright/data/media_codecs_sw.xml
@@ -203,8 +203,7 @@
<Feature name="dynamic-color-aspects" />
<Attribute name="software-codec" />
</MediaCodec>
- <MediaCodec name="c2.android.av1.decoder" type="video/av01" variant="slow-cpu,!slow-cpu">
- <!-- TODO: implement a mechanism to prevent AV1 Decoder usage on pre-U devices -->
+ <MediaCodec name="c2.android.av1-dav1d.decoder" type="video/av01" variant="slow-cpu,!slow-cpu">
<Limit name="alignment" value="1x1" />
<Limit name="block-size" value="16x16" />
<Variant name="!slow-cpu">
@@ -224,7 +223,8 @@
<Feature name="low-latency" />
<Attribute name="software-codec" />
</MediaCodec>
- <MediaCodec name="c2.android.av1-dav1d.decoder" type="video/av01" variant="slow-cpu,!slow-cpu" rank="1024">
+ <MediaCodec name="c2.android.av1.decoder" type="video/av01" variant="slow-cpu,!slow-cpu" rank="1024">
+ <!-- TODO: implement a mechanism to prevent AV1 Decoder usage on pre-U devices -->
<Limit name="alignment" value="1x1" />
<Limit name="block-size" value="16x16" />
<Variant name="!slow-cpu">
diff --git a/media/module/extractors/mp4/MPEG4Extractor.cpp b/media/module/extractors/mp4/MPEG4Extractor.cpp
index cb2994e..b3707c8 100644
--- a/media/module/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/module/extractors/mp4/MPEG4Extractor.cpp
@@ -1615,6 +1615,39 @@
mLastTrack->timescale = ntohl(timescale);
+ // 14496-12 says all ones means indeterminate, but some files seem to use
+ // 0 instead. We treat both the same.
+ int64_t duration = 0;
+ if (version == 1) {
+ if (mDataSource->readAt(
+ timescale_offset + 4, &duration, sizeof(duration))
+ < (ssize_t)sizeof(duration)) {
+ return ERROR_IO;
+ }
+ if (duration != -1) {
+ duration = ntoh64(duration);
+ }
+ } else {
+ uint32_t duration32;
+ if (mDataSource->readAt(
+ timescale_offset + 4, &duration32, sizeof(duration32))
+ < (ssize_t)sizeof(duration32)) {
+ return ERROR_IO;
+ }
+ if (duration32 != 0xffffffff) {
+ duration = ntohl(duration32);
+ }
+ }
+ if (duration != 0 && mLastTrack->timescale != 0) {
+ long double durationUs = ((long double)duration * 1000000) / mLastTrack->timescale;
+ if (durationUs < 0 || durationUs > INT64_MAX) {
+ ALOGE("cannot represent %lld * 1000000 / %lld in 64 bits",
+ (long long) duration, (long long) mLastTrack->timescale);
+ return ERROR_MALFORMED;
+ }
+ AMediaFormat_setInt64(mLastTrack->meta, AMEDIAFORMAT_KEY_DURATION, durationUs);
+ }
+
uint8_t lang[2];
off64_t lang_offset;
if (version == 1) {
@@ -3874,18 +3907,17 @@
}
int32_t id;
- int64_t duration;
if (version == 1) {
// we can get ctime value from U64_AT(&buffer[4])
// we can get mtime value from U64_AT(&buffer[12])
id = U32_AT(&buffer[20]);
- duration = U64_AT(&buffer[28]);
+ // we can get duration value from U64_AT(&buffer[28])
} else if (version == 0) {
// we can get ctime value from U32_AT(&buffer[4])
// we can get mtime value from U32_AT(&buffer[8])
id = U32_AT(&buffer[12]);
- duration = U32_AT(&buffer[20]);
+ // we can get duration value from U32_AT(&buffer[20])
} else {
return ERROR_UNSUPPORTED;
}
@@ -3894,15 +3926,6 @@
return ERROR_MALFORMED;
AMediaFormat_setInt32(mLastTrack->meta, AMEDIAFORMAT_KEY_TRACK_ID, id);
- if (duration != 0 && mHeaderTimescale != 0) {
- long double durationUs = ((long double)duration * 1000000) / mHeaderTimescale;
- if (durationUs < 0 || durationUs > INT64_MAX) {
- ALOGE("cannot represent %lld * 1000000 / %lld in 64 bits",
- (long long) duration, (long long) mHeaderTimescale);
- return ERROR_MALFORMED;
- }
- AMediaFormat_setInt64(mLastTrack->meta, AMEDIAFORMAT_KEY_DURATION, durationUs);
- }
size_t matrixOffset = dynSize + 16;
int32_t a00 = U32_AT(&buffer[matrixOffset]);
diff --git a/media/ndk/Android.bp b/media/ndk/Android.bp
index 9ec7700..3d873df 100644
--- a/media/ndk/Android.bp
+++ b/media/ndk/Android.bp
@@ -192,7 +192,6 @@
header_libs: [
"libstagefright_headers",
"libmedia_headers",
- "libstagefright_headers",
],
shared_libs: [
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index d5d778f..af67a56 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1935,10 +1935,11 @@
if (mPrimaryHardwareDev == nullptr) {
return 0;
}
+ if (mInputBufferSizeOrderedDevs.empty()) {
+ return 0;
+ }
mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
- sp<DeviceHalInterface> dev = mPrimaryHardwareDev.load()->hwDevice();
-
std::vector<audio_channel_mask_t> channelMasks = {channelMask};
if (channelMask != AUDIO_CHANNEL_IN_MONO) {
channelMasks.push_back(AUDIO_CHANNEL_IN_MONO);
@@ -1968,6 +1969,22 @@
mHardwareStatus = AUDIO_HW_IDLE;
+ auto getInputBufferSize = [](const sp<DeviceHalInterface>& dev, audio_config_t config,
+ size_t* bytes) -> status_t {
+ if (!dev) {
+ return BAD_VALUE;
+ }
+ status_t result = dev->getInputBufferSize(&config, bytes);
+ if (result == BAD_VALUE) {
+ // Retry with the config suggested by the HAL.
+ result = dev->getInputBufferSize(&config, bytes);
+ }
+ if (result != OK || *bytes == 0) {
+ return BAD_VALUE;
+ }
+ return result;
+ };
+
// Change parameters of the configuration each iteration until we find a
// configuration that the device will support, or HAL suggests what it supports.
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
@@ -1979,16 +1996,15 @@
config.sample_rate = testSampleRate;
size_t bytes = 0;
- audio_config_t loopConfig = config;
- status_t result = dev->getInputBufferSize(&config, &bytes);
- if (result == BAD_VALUE) {
- // Retry with the config suggested by the HAL.
- result = dev->getInputBufferSize(&config, &bytes);
+ ret = BAD_VALUE;
+ for (const AudioHwDevice* dev : mInputBufferSizeOrderedDevs) {
+ ret = getInputBufferSize(dev->hwDevice(), config, &bytes);
+ if (ret == OK) {
+ break;
+ }
}
- if (result != OK || bytes == 0) {
- config = loopConfig;
- continue;
- }
+ if (ret == BAD_VALUE) continue;
+
if (config.sample_rate != sampleRate || config.channel_mask != channelMask ||
config.format != format) {
uint32_t dstChannelCount = audio_channel_count_from_in_mask(channelMask);
@@ -2610,12 +2626,43 @@
}
mAudioHwDevs.add(handle, audioDevice);
+ if (strcmp(name, AUDIO_HARDWARE_MODULE_ID_STUB) != 0) {
+ mInputBufferSizeOrderedDevs.insert(audioDevice);
+ }
ALOGI("loadHwModule() Loaded %s audio interface, handle %d", name, handle);
return audioDevice;
}
+// Sort AudioHwDevice to be traversed in the getInputBufferSize call in the following order:
+// Primary, Usb, Bluetooth, A2DP, other modules, remote submix.
+/* static */
+bool AudioFlinger::inputBufferSizeDevsCmp(const AudioHwDevice* lhs, const AudioHwDevice* rhs) {
+ static const std::map<std::string_view, int> kPriorities = {
+ { AUDIO_HARDWARE_MODULE_ID_PRIMARY, 0 }, { AUDIO_HARDWARE_MODULE_ID_USB, 1 },
+ { AUDIO_HARDWARE_MODULE_ID_BLUETOOTH, 2 }, { AUDIO_HARDWARE_MODULE_ID_A2DP, 3 },
+ { AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, std::numeric_limits<int>::max() }
+ };
+
+ const std::string_view lhsName = lhs->moduleName();
+ const std::string_view rhsName = rhs->moduleName();
+
+ auto lhsPriority = std::numeric_limits<int>::max() - 1;
+ if (const auto lhsIt = kPriorities.find(lhsName); lhsIt != kPriorities.end()) {
+ lhsPriority = lhsIt->second;
+ }
+ auto rhsPriority = std::numeric_limits<int>::max() - 1;
+ if (const auto rhsIt = kPriorities.find(rhsName); rhsIt != kPriorities.end()) {
+ rhsPriority = rhsIt->second;
+ }
+
+ if (lhsPriority != rhsPriority) {
+ return lhsPriority < rhsPriority;
+ }
+ return lhsName < rhsName;
+}
+
// ----------------------------------------------------------------------------
uint32_t AudioFlinger::getPrimaryOutputSamplingRate() const
@@ -4915,6 +4962,23 @@
return mPatchPanel->getAudioMixPort_l(devicePort, mixPort);
}
+status_t AudioFlinger::setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& tracksInternalMute) {
+ audio_utils::lock_guard _l(mutex());
+ ALOGV("%s", __func__);
+
+ std::map<audio_port_handle_t, bool> tracksInternalMuteMap;
+ for (const auto& trackInternalMute : tracksInternalMute) {
+ audio_port_handle_t portId = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_int32_t_audio_port_handle_t(trackInternalMute.portId));
+ tracksInternalMuteMap.emplace(portId, trackInternalMute.muted);
+ }
+ for (size_t i = 0; i < mPlaybackThreads.size() && !tracksInternalMuteMap.empty(); i++) {
+ mPlaybackThreads.valueAt(i)->setTracksInternalMute(&tracksInternalMuteMap);
+ }
+ return NO_ERROR;
+}
+
// ----------------------------------------------------------------------------
status_t AudioFlinger::onTransactWrapper(TransactionCode code,
@@ -4949,6 +5013,7 @@
case TransactionCode::INVALIDATE_TRACKS:
case TransactionCode::GET_AUDIO_POLICY_CONFIG:
case TransactionCode::GET_AUDIO_MIX_PORT:
+ case TransactionCode::SET_TRACKS_INTERNAL_MUTE:
ALOGW("%s: transaction %d received from PID %d",
__func__, static_cast<int>(code), IPCThreadState::self()->getCallingPid());
// return status only for non void methods
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 4e46bea..4711dc5 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -259,6 +259,10 @@
status_t getAudioMixPort(const struct audio_port_v7* devicePort,
struct audio_port_v7* mixPort) const final EXCLUDES_AudioFlinger_Mutex;
+ status_t setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& tracksInternalMute) final
+ EXCLUDES_AudioFlinger_Mutex;
+
status_t onTransactWrapper(TransactionCode code, const Parcel& data, uint32_t flags,
const std::function<status_t()>& delegate) final EXCLUDES_AudioFlinger_Mutex;
@@ -630,6 +634,10 @@
DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs
GUARDED_BY(hardwareMutex()) {nullptr /* defValue */};
+ static bool inputBufferSizeDevsCmp(const AudioHwDevice* lhs, const AudioHwDevice* rhs);
+ std::set<AudioHwDevice*, decltype(&inputBufferSizeDevsCmp)>
+ mInputBufferSizeOrderedDevs GUARDED_BY(hardwareMutex()) {inputBufferSizeDevsCmp};
+
const sp<DevicesFactoryHalInterface> mDevicesFactoryHal =
DevicesFactoryHalInterface::create();
/* const */ sp<DevicesFactoryHalCallback> mDevicesFactoryHalCallback; // set onFirstRef().
diff --git a/services/audioflinger/IAfThread.h b/services/audioflinger/IAfThread.h
index 4b6ab89..204cc63 100644
--- a/services/audioflinger/IAfThread.h
+++ b/services/audioflinger/IAfThread.h
@@ -536,6 +536,9 @@
virtual const std::atomic<int64_t>& framesWritten() const = 0;
virtual bool usesHwAvSync() const = 0;
+
+ virtual void setTracksInternalMute(std::map<audio_port_handle_t, bool>* tracksInternalMute)
+ EXCLUDES_ThreadBase_Mutex = 0;
};
class IAfDirectOutputThread : public virtual IAfPlaybackThread {
diff --git a/services/audioflinger/IAfTrack.h b/services/audioflinger/IAfTrack.h
index a0d20ac..168bcca 100644
--- a/services/audioflinger/IAfTrack.h
+++ b/services/audioflinger/IAfTrack.h
@@ -427,6 +427,10 @@
virtual FillingStatus& fillingStatus() = 0;
virtual int8_t& retryCount() = 0;
virtual FastTrackUnderruns& fastTrackUnderruns() = 0;
+
+ // Internal mute, this is currently only used for bit-perfect playback
+ virtual bool getInternalMute() const = 0;
+ virtual void setInternalMute(bool muted) = 0;
};
// playback track, used by DuplicatingThread
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 89ee5c0..1e1b6d2 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -220,6 +220,8 @@
*/
void processMuteEvent_l(const sp<IAudioManager>& audioManager, mute_state_t muteState) final;
+ bool getInternalMute() const final { return mInternalMute; }
+ void setInternalMute(bool muted) final { mInternalMute = muted; }
protected:
DISALLOW_COPY_AND_ASSIGN(Track);
@@ -401,6 +403,8 @@
// access these two variables only when holding player thread lock.
std::unique_ptr<os::PersistableBundle> mMuteEventExtras;
mute_state_t mMuteState;
+
+ bool mInternalMute = false;
}; // end of Track
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 7c248dc..d59697d 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -33,6 +33,7 @@
#include <afutils/Vibrator.h>
#include <audio_utils/MelProcessor.h>
#include <audio_utils/Metadata.h>
+#include <com_android_media_audioserver.h>
#ifdef DEBUG_CPU_USAGE
#include <audio_utils/Statistics.h>
#include <cpustats/ThreadCpuUsage.h>
@@ -2696,14 +2697,17 @@
}
}
- // Set DIRECT flag if current thread is DirectOutputThread. This can
- // happen when the playback is rerouted to direct output thread by
+ // Set DIRECT/OFFLOAD flag if current thread is DirectOutputThread/OffloadThread.
+ // This can happen when the playback is rerouted to direct output/offload thread by
// dynamic audio policy.
// Do NOT report the flag changes back to client, since the client
- // doesn't explicitly request a direct flag.
+ // doesn't explicitly request a direct/offload flag.
audio_output_flags_t trackFlags = *flags;
if (mType == DIRECT) {
trackFlags = static_cast<audio_output_flags_t>(trackFlags | AUDIO_OUTPUT_FLAG_DIRECT);
+ } else if (mType == OFFLOAD) {
+ trackFlags = static_cast<audio_output_flags_t>(trackFlags |
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT);
}
*afTrackFlags = trackFlags;
@@ -5771,6 +5775,11 @@
vlf *= volume;
vrf *= volume;
+ if (track->getInternalMute()) {
+ vlf = 0.f;
+ vrf = 0.f;
+ }
+
track->setFinalVolume(vlf, vrf);
++fastTracks;
} else {
@@ -5970,6 +5979,11 @@
vaf = v * sendLevel * (1. / MAX_GAIN_INT);
}
+ if (track->getInternalMute()) {
+ vrf = 0.f;
+ vlf = 0.f;
+ }
+
track->setFinalVolume(vlf, vrf);
// Delegate volume control to effect in track effect chain if needed
@@ -11353,14 +11367,15 @@
// If there is only one active track and it is bit-perfect, enable tee buffer.
float volumeLeft = 1.0f;
float volumeRight = 1.0f;
- if (mActiveTracks.size() == 1 && mActiveTracks[0]->isBitPerfect()) {
- const int trackId = mActiveTracks[0]->id();
+ if (sp<IAfTrack> bitPerfectTrack = getTrackToStreamBitPerfectly_l();
+ bitPerfectTrack != nullptr) {
+ const int trackId = bitPerfectTrack->id();
mAudioMixer->setParameter(
trackId, AudioMixer::TRACK, AudioMixer::TEE_BUFFER, (void *)mSinkBuffer);
mAudioMixer->setParameter(
trackId, AudioMixer::TRACK, AudioMixer::TEE_BUFFER_FRAME_COUNT,
(void *)(uintptr_t)mNormalFrameCount);
- mActiveTracks[0]->getFinalVolume(&volumeLeft, &volumeRight);
+ bitPerfectTrack->getFinalVolume(&volumeLeft, &volumeRight);
mIsBitPerfect = true;
} else {
mIsBitPerfect = false;
@@ -11385,4 +11400,39 @@
mHasDataCopiedToSinkBuffer = mIsBitPerfect;
}
+void BitPerfectThread::setTracksInternalMute(
+ std::map<audio_port_handle_t, bool>* tracksInternalMute) {
+ for (auto& track : mTracks) {
+ if (auto it = tracksInternalMute->find(track->portId()); it != tracksInternalMute->end()) {
+ track->setInternalMute(it->second);
+ tracksInternalMute->erase(it);
+ }
+ }
+}
+
+sp<IAfTrack> BitPerfectThread::getTrackToStreamBitPerfectly_l() {
+ if (com::android::media::audioserver::
+ fix_concurrent_playback_behavior_with_bit_perfect_client()) {
+ sp<IAfTrack> bitPerfectTrack = nullptr;
+ bool allOtherTracksMuted = true;
+ // Return the bit perfect track if all other tracks are muted
+ for (const auto& track : mActiveTracks) {
+ if (track->isBitPerfect()) {
+ bitPerfectTrack = track;
+ } else if (track->getFinalVolume() != 0.f) {
+ allOtherTracksMuted = false;
+ if (bitPerfectTrack != nullptr) {
+ break;
+ }
+ }
+ }
+ return allOtherTracksMuted ? bitPerfectTrack : nullptr;
+ } else {
+ if (mActiveTracks.size() == 1 && mActiveTracks[0]->isBitPerfect()) {
+ return mActiveTracks[0];
+ }
+ }
+ return nullptr;
+}
+
} // namespace android
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 86e1894..7e9bef1 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1200,6 +1200,11 @@
}
return mHalStarted;
}
+
+ void setTracksInternalMute(std::map<audio_port_handle_t, bool>* /* tracksInternalMute */)
+ override EXCLUDES_ThreadBase_Mutex {
+ // Do nothing. It is only used for bit perfect thread
+ }
protected:
// updated by readOutputParameters_l()
size_t mNormalFrameCount; // normal mixer and effects
@@ -2449,12 +2454,17 @@
BitPerfectThread(const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut *output,
audio_io_handle_t id, bool systemReady);
+ void setTracksInternalMute(std::map<audio_port_handle_t, bool>* tracksInternalMuted)
+ final EXCLUDES_ThreadBase_Mutex;
+
protected:
mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) final
REQUIRES(mutex(), ThreadBase_ThreadLoop);
void threadLoop_mix() final REQUIRES(ThreadBase_ThreadLoop);
private:
+ sp<IAfTrack> getTrackToStreamBitPerfectly_l() REQUIRES(mutex());
+
// These variables are only accessed on the threadLoop; hence need no mutex.
bool mIsBitPerfect GUARDED_BY(ThreadBase_ThreadLoop) = false;
float mVolumeLeft GUARDED_BY(ThreadBase_ThreadLoop) = 0.f;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index c40f795..f8cfc12 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -913,7 +913,7 @@
" Format Chn mask SRate "
"ST Usg CT "
" G db L dB R dB VS dB "
- " Server FrmCnt FrmRdy F Underruns Flushed BitPerfect"
+ " Server FrmCnt FrmRdy F Underruns Flushed BitPerfect InternalMute"
"%s\n",
isServerLatencySupported() ? " Latency" : "");
}
@@ -999,7 +999,7 @@
"%08X %08X %6u "
"%2u %3x %2x "
"%5.2g %5.2g %5.2g %5.2g%c "
- "%08X %6zu%c %6zu %c %9u%c %7u %10s",
+ "%08X %6zu%c %6zu %c %9u%c %7u %10s %12s",
active ? "yes" : "no",
(mClient == 0) ? getpid() : mClient->pid(),
mSessionId,
@@ -1029,7 +1029,8 @@
mAudioTrackServerProxy->getUnderrunFrames(),
nowInUnderrun,
(unsigned)mAudioTrackServerProxy->framesFlushed() % 10000000,
- isBitPerfect() ? "true" : "false"
+ isBitPerfect() ? "true" : "false",
+ getInternalMute() ? "true" : "false"
);
if (isServerLatencySupported()) {
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 8f17ffc..de6eed8 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -18,6 +18,7 @@
#define ANDROID_AUDIOPOLICY_INTERFACE_H
#include <android/media/DeviceConnectedState.h>
+#include <android/media/TrackInternalMuteInfo.h>
#include <media/AudioCommonTypes.h>
#include <media/AudioContainers.h>
#include <media/AudioDeviceTypeAddr.h>
@@ -585,6 +586,9 @@
// Get the attributes of the mix port when connecting to the given device port.
virtual status_t getAudioMixPort(const struct audio_port_v7 *devicePort,
struct audio_port_v7 *mixPort) = 0;
+
+ virtual status_t setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& tracksInternalMute) = 0;
};
// These are the signatures of createAudioPolicyManager/destroyAudioPolicyManager
diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h
index 4643bd1..04d5362 100644
--- a/services/audiopolicy/common/include/policy.h
+++ b/services/audiopolicy/common/include/policy.h
@@ -17,6 +17,7 @@
#pragma once
#include <system/audio.h>
+#include <set>
#include <vector>
#include <media/AudioContainers.h>
@@ -27,6 +28,10 @@
static const audio_attributes_t defaultAttr = AUDIO_ATTRIBUTES_INITIALIZER;
+static const std::set<audio_usage_t > gHighPriorityUseCases = {
+ AUDIO_USAGE_ALARM, AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE
+};
+
} // namespace android
static const audio_format_t gDynamicFormat = AUDIO_FORMAT_DEFAULT;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index 00958aa..914f3fe 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -29,6 +29,7 @@
#include "ClientDescriptor.h"
#include "DeviceDescriptor.h"
#include "PolicyAudioPort.h"
+#include "PreferredMixerAttributesInfo.h"
#include <vector>
namespace android {
@@ -477,6 +478,16 @@
PortHandleVector getClientsForStream(audio_stream_type_t streamType) const;
+ bool isBitPerfect() const {
+ return (getFlags().output & AUDIO_OUTPUT_FLAG_BIT_PERFECT) != AUDIO_OUTPUT_FLAG_NONE;
+ }
+
+ /**
+ * Return true if there is any client with the same usage active on the given device.
+ * When the given device is null, return true if there is any client active.
+ */
+ bool isUsageActiveOnDevice(audio_usage_t usage, sp<DeviceDescriptor> device) const;
+
virtual std::string info() const override;
const sp<IOProfile> mProfile; // I/O profile this output derives from
@@ -489,7 +500,7 @@
audio_session_t mDirectClientSession; // session id of the direct output client
bool mPendingReopenToQueryProfiles = false;
audio_channel_mask_t mMixerChannelMask = AUDIO_CHANNEL_NONE;
- bool mUsePreferredMixerAttributes = false;
+ sp<PreferredMixerAttributesInfo> mPreferredAttrInfo = nullptr;
};
// Audio output driven by an input device directly.
@@ -616,6 +627,8 @@
*/
bool isAnyDeviceTypeActive(const DeviceTypeSet& deviceTypes) const;
+ bool isUsageActiveOnDevice(audio_usage_t usage, sp<DeviceDescriptor> device) const;
+
void dump(String8 *dst) const;
};
diff --git a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
index fe90a1e..a596c43 100644
--- a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
@@ -165,6 +165,18 @@
mIsInvalid = true;
}
+ bool getInternalMute() const { return mInternalMute; }
+
+ /**
+ * Set the internal mute for a client. Return true if the existing value is different from
+ * the given value.
+ */
+ bool setInternalMute(bool muted) {
+ const bool result = (mInternalMute != muted);
+ mInternalMute = muted;
+ return result;
+ }
+
private:
const audio_stream_type_t mStream;
const product_strategy_t mStrategy;
@@ -178,6 +190,7 @@
*/
uint32_t mActivityCount = 0;
bool mIsInvalid = false;
+ bool mInternalMute = false;
};
class RecordClientDescriptor: public ClientDescriptor
diff --git a/services/audiopolicy/common/managerdefinitions/include/PreferredMixerAttributesInfo.h b/services/audiopolicy/common/managerdefinitions/include/PreferredMixerAttributesInfo.h
index 9472481..a493e3c 100644
--- a/services/audiopolicy/common/managerdefinitions/include/PreferredMixerAttributesInfo.h
+++ b/services/audiopolicy/common/managerdefinitions/include/PreferredMixerAttributesInfo.h
@@ -44,6 +44,17 @@
void increaseActiveClient() { mActiveClientsCount++; }
void decreaseActiveClient() { mActiveClientsCount--; }
+ void resetActiveClient() { mActiveClientsCount = 0; }
+
+ bool isBitPerfect() const {
+ return (getFlags() & AUDIO_OUTPUT_FLAG_BIT_PERFECT) != AUDIO_OUTPUT_FLAG_NONE;
+ }
+
+ bool configMatches(const audio_config_t& config) const {
+ return config.format == mMixerAttributes.config.format &&
+ config.channel_mask == mMixerAttributes.config.channel_mask &&
+ config.sample_rate == mMixerAttributes.config.sample_rate;
+ }
void dump(String8 *dst);
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 6537a00..68b2e7b 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -561,6 +561,7 @@
audio_port_config config = {};
devicePort->toAudioPortConfig(&config);
config.config_mask = AUDIO_PORT_CONFIG_GAIN;
+ config.gain.mode = gains[0]->getMode();
config.gain.values[0] = gainValueMb;
return mClientInterface->setAudioPortConfig(&config, 0) == NO_ERROR;
}
@@ -791,6 +792,16 @@
mDevices = devices;
}
+bool SwAudioOutputDescriptor::isUsageActiveOnDevice(audio_usage_t usage,
+ sp<android::DeviceDescriptor> device) const {
+ if (device != nullptr && !mDevices.contains(device)) {
+ return false;
+ }
+ return std::any_of(mActiveClients.begin(), mActiveClients.end(),
+ [usage](sp<TrackClientDescriptor> client) {
+ return client->attributes().usage == usage; });
+}
+
// HwAudioOutputDescriptor implementation
HwAudioOutputDescriptor::HwAudioOutputDescriptor(const sp<SourceClientDescriptor>& source,
AudioPolicyClientInterface *clientInterface)
@@ -1016,6 +1027,17 @@
return clientsForStream;
}
+bool SwAudioOutputCollection::isUsageActiveOnDevice(audio_usage_t usage,
+ sp<android::DeviceDescriptor> device) const {
+ for (size_t i = 0; i < this->size(); i++) {
+ const sp<SwAudioOutputDescriptor> outputDesc = this->valueAt(i);
+ if (outputDesc->isUsageActiveOnDevice(usage, device)) {
+ return true;
+ }
+ }
+ return false;
+}
+
std::string SwAudioOutputDescriptor::info() const {
std::string result;
result.append("[" );
diff --git a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
index 2aee501..667c189 100644
--- a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
@@ -57,8 +57,8 @@
void TrackClientDescriptor::dump(String8 *dst, int spaces) const
{
ClientDescriptor::dump(dst, spaces);
- dst->appendFormat("%*sStream: %d; Flags: %08x; Refcount: %d\n", spaces, "",
- mStream, mFlags, mActivityCount);
+ dst->appendFormat("%*sStream: %d; Flags: %08x; Refcount: %d; InternalMute: %s\n",
+ spaces, "", mStream, mFlags, mActivityCount, mInternalMute ? "Yes" : "No");
dst->appendFormat("%*sDAP Primary Mix: %p\n", spaces, "", mPrimaryMix.promote().get());
if (!mSecondaryOutputs.empty()) {
dst->appendFormat("%*sDAP Secondary Outputs: ", spaces - 2, "");
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 7f4be79..3e693ec 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -338,7 +338,7 @@
&& (!device_distinguishes_on_address(device->type())
// always force when disconnecting (a non-duplicated device)
|| (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
- if (desc->mUsePreferredMixerAttributes && newDevices != desc->devices()) {
+ if (desc->mPreferredAttrInfo != nullptr && newDevices != desc->devices()) {
// If the device is using preferred mixer attributes, the output need to reopen
// with default configuration when the new selected devices are different from
// current routing devices
@@ -917,15 +917,15 @@
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
DeviceVector newDevices = getNewOutputDevices(desc, true /*fromCache*/);
+ if (state != AUDIO_MODE_NORMAL && oldState == AUDIO_MODE_NORMAL
+ && desc->mPreferredAttrInfo != nullptr) {
+ // If the output is using preferred mixer attributes and the audio mode is not normal,
+ // the output need to reopen with default configuration.
+ outputsToReopen.emplace(mOutputs.keyAt(i), newDevices);
+ continue;
+ }
if (state != AUDIO_MODE_IN_CALL || (desc != mPrimaryOutput && !isTelephonyRxOrTx(desc))) {
bool forceRouting = !newDevices.isEmpty();
- if (desc->mUsePreferredMixerAttributes && newDevices != desc->devices()) {
- // If the device is using preferred mixer attributes, the output need to reopen
- // with default configuration when the new selected devices are different from
- // current routing devices.
- outputsToReopen.emplace(mOutputs.keyAt(i), newDevices);
- continue;
- }
setOutputDevices(__func__, desc, newDevices, forceRouting, 0 /*delayMs*/, nullptr,
true /*requiresMuteCheck*/, !forceRouting /*requiresVolumeCheck*/);
}
@@ -1337,23 +1337,40 @@
// Only use preferred mixer if the uid matches or the preferred mixer is bit-perfect
// and it is currently active.
if (info != nullptr && info->getUid() != uid &&
- ((info->getFlags() & AUDIO_OUTPUT_FLAG_BIT_PERFECT) == AUDIO_OUTPUT_FLAG_NONE ||
- info->getActiveClientCount() == 0)) {
+ (!info->isBitPerfect() || info->getActiveClientCount() == 0)) {
info = nullptr;
}
+ if (com::android::media::audioserver::
+ fix_concurrent_playback_behavior_with_bit_perfect_client()) {
+ if (info != nullptr && info->getUid() == uid &&
+ info->configMatches(*config) &&
+ (mEngine->getPhoneState() != AUDIO_MODE_NORMAL ||
+ std::any_of(gHighPriorityUseCases.begin(), gHighPriorityUseCases.end(),
+ [this, &outputDevices](audio_usage_t usage) {
+ return mOutputs.isUsageActiveOnDevice(
+ usage, outputDevices[0]); }))) {
+ // Bit-perfect request is not allowed when the phone mode is not normal or
+ // there is any higher priority user case active.
+ return INVALID_OPERATION;
+ }
+ }
}
*output = getOutputForDevices(outputDevices, session, resultAttr, config,
flags, isSpatialized, info, resultAttr->flags & AUDIO_FLAG_MUTE_HAPTIC);
// The client will be active if the client is currently preferred mixer owner and the
// requested configuration matches the preferred mixer configuration.
*isBitPerfect = (info != nullptr
- && (info->getFlags() & AUDIO_OUTPUT_FLAG_BIT_PERFECT) != AUDIO_OUTPUT_FLAG_NONE
+ && info->isBitPerfect()
&& info->getUid() == uid
&& *output != AUDIO_IO_HANDLE_NONE
// When bit-perfect output is selected for the preferred mixer attributes owner,
// only need to consider the config matches.
&& mOutputs.valueFor(*output)->isConfigurationMatched(
clientConfig, AUDIO_OUTPUT_FLAG_NONE));
+
+ if (*isBitPerfect) {
+ *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_BIT_PERFECT);
+ }
}
if (*output == AUDIO_IO_HANDLE_NONE) {
AudioProfileVector profiles;
@@ -1708,6 +1725,24 @@
// at this stage we should ignore the DIRECT flag as no direct output could be
// found earlier
*flags = (audio_output_flags_t) (*flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
+ if (com::android::media::audioserver::
+ fix_concurrent_playback_behavior_with_bit_perfect_client()) {
+ // If the preferred mixer attributes is null, do not select the bit-perfect output
+ // unless the bit-perfect output is the only output.
+ // The bit-perfect output can exist while the passed in preferred mixer attributes
+ // info is null when it is a high priority client. The high priority clients are
+ // ringtone or alarm, which is not a bit-perfect use case.
+ size_t i = 0;
+ while (i < outputs.size() && outputs.size() > 1) {
+ auto desc = mOutputs.valueFor(outputs[i]);
+ // The output descriptor must not be null here.
+ if (desc->isBitPerfect()) {
+ outputs.removeItemsAt(i);
+ } else {
+ i += 1;
+ }
+ }
+ }
output = selectOutput(
outputs, *flags, config->format, channelMask, config->sample_rate, session);
}
@@ -2191,6 +2226,20 @@
ALOGV("startOutput() output %d, stream %d, session %d",
outputDesc->mIoHandle, client->stream(), client->session());
+ if (com::android::media::audioserver::fix_concurrent_playback_behavior_with_bit_perfect_client()
+ && gHighPriorityUseCases.count(client->attributes().usage) != 0
+ && outputDesc->isBitPerfect()) {
+ // Usually, APM selects bit-perfect output for high priority use cases only when
+ // bit-perfect output is the only output that can be routed to the selected device.
+ // However, here is no need to play high priority use cases such as ringtone and alarm
+ // on the bit-perfect path. Reopen the output and return DEAD_OBJECT so that the client
+ // can attach to new output.
+ ALOGD("%s: reopen bit-perfect output as high priority use case(%d) is starting",
+ __func__, client->stream());
+ reopenOutput(outputDesc, nullptr /*config*/, AUDIO_OUTPUT_FLAG_NONE, __func__);
+ return DEAD_OBJECT;
+ }
+
status_t status = outputDesc->start();
if (status != NO_ERROR) {
return status;
@@ -2209,7 +2258,6 @@
ALOGE("%s unable to open output with default config", __func__);
return status;
}
- desc->mUsePreferredMixerAttributes = true;
}
return status;
}
@@ -2233,14 +2281,13 @@
if (desc == nullptr) {
return BAD_VALUE;
}
- desc->mUsePreferredMixerAttributes = true;
+ desc->mPreferredAttrInfo = info;
// Intentionally return error to let the client side resending request for
// creating and starting.
return DEAD_OBJECT;
}
info->increaseActiveClient();
- if (info->getActiveClientCount() == 1 &&
- (info->getFlags() & AUDIO_OUTPUT_FLAG_BIT_PERFECT) != AUDIO_OUTPUT_FLAG_NONE) {
+ if (info->getActiveClientCount() == 1 && info->isBitPerfect()) {
// If it is first bit-perfect client, reroute all clients that will be routed to
// the bit-perfect sink so that it is guaranteed only bit-perfect stream is active.
PortHandleVector clientsToInvalidate;
@@ -2270,6 +2317,15 @@
usleep(delayMs * 1000);
}
+ if (status == NO_ERROR &&
+ outputDesc->mPreferredAttrInfo != nullptr &&
+ outputDesc->isBitPerfect() &&
+ com::android::media::audioserver::
+ fix_concurrent_playback_behavior_with_bit_perfect_client()) {
+ // A new client is started on bit-perfect output, update all clients internal mute.
+ updateClientsInternalMute(outputDesc);
+ }
+
return status;
}
@@ -2374,6 +2430,11 @@
followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_NOTIFICATION)) ||
(beaconMuteLatency > 0));
uint32_t waitMs = beaconMuteLatency;
+ const bool needToCloseBitPerfectOutput =
+ (com::android::media::audioserver::
+ fix_concurrent_playback_behavior_with_bit_perfect_client() &&
+ gHighPriorityUseCases.count(clientAttr.usage) != 0);
+ std::vector<sp<SwAudioOutputDescriptor>> outputsToReopen;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != outputDesc) {
@@ -2410,15 +2471,22 @@
// Note restoring AudioTracks onto this output needs to invoke
// a volume ramp if there is no mute.
requiresMuteCheck |= sharedDevice && isActive;
+
+ if (needToCloseBitPerfectOutput && desc->isBitPerfect()) {
+ outputsToReopen.push_back(desc);
+ }
}
}
- if (outputDesc->mUsePreferredMixerAttributes && devices != outputDesc->devices()) {
+ if (outputDesc->mPreferredAttrInfo != nullptr && devices != outputDesc->devices()) {
// If the output is open with preferred mixer attributes, but the routed device is
// changed when calling this function, returning DEAD_OBJECT to indicate routing
// changed.
return DEAD_OBJECT;
}
+ for (auto& outputToReopen : outputsToReopen) {
+ reopenOutput(outputToReopen, nullptr /*config*/, AUDIO_OUTPUT_FLAG_NONE, __func__);
+ }
const uint32_t muteWaitMs =
setOutputDevices(__func__, outputDesc, devices, force, 0, nullptr,
requiresMuteCheck);
@@ -2497,7 +2565,7 @@
getAudioDeviceOutLeAudioUnicastSet()).isEmpty()))) {
DeviceVector newDevices = getNewOutputDevices(desc, false /*fromCache*/);
bool force = desc->devices() != newDevices;
- if (desc->mUsePreferredMixerAttributes && force) {
+ if (desc->mPreferredAttrInfo != nullptr && force) {
// If the device is using preferred mixer attributes, the output need to reopen
// with default configuration when the new selected devices are different from
// current routing devices.
@@ -2545,12 +2613,21 @@
if (outputDesc->devices().size() == 1) {
sp<PreferredMixerAttributesInfo> info = getPreferredMixerAttributesInfo(
outputDesc->devices()[0]->getId(), client->strategy());
+ bool outputReopened = false;
if (info != nullptr && info->getUid() == client->uid()) {
info->decreaseActiveClient();
if (info->getActiveClientCount() == 0) {
reopenOutput(outputDesc, nullptr /*config*/, AUDIO_OUTPUT_FLAG_NONE, __func__);
+ outputReopened = true;
}
}
+ if (com::android::media::audioserver::
+ fix_concurrent_playback_behavior_with_bit_perfect_client() &&
+ !outputReopened && outputDesc->isBitPerfect()) {
+ // Only need to update the clients' internal mute when the output is bit-perfect and it
+ // is not reopened.
+ updateClientsInternalMute(outputDesc);
+ }
}
return status;
}
@@ -2622,7 +2699,7 @@
DeviceVector newDevices2 = getNewOutputDevices(desc, false /*fromCache*/);
bool force = desc->devices() != newDevices2;
- if (desc->mUsePreferredMixerAttributes && force) {
+ if (desc->mPreferredAttrInfo != nullptr && force) {
// If the device is using preferred mixer attributes, the output need to
// reopen with default configuration when the new selected devices are
// different from current routing devices.
@@ -4114,7 +4191,7 @@
// preventing the force re-routing in case of default dev that distinguishes on address.
// Let's give back to engine full device choice decision however.
bool forceRouting = !newDevices.isEmpty();
- if (outputDesc->mUsePreferredMixerAttributes && newDevices != outputDesc->devices()) {
+ if (outputDesc->mPreferredAttrInfo != nullptr && newDevices != outputDesc->devices()) {
// If the device is using preferred mixer attributes, the output need to reopen
// with default configuration when the new selected devices are different from
// current routing devices.
@@ -4715,7 +4792,7 @@
const auto output = mOutputs.valueAt(i);
if (output->mProfile == profile && output->devices().onlyContainsDevice(deviceDescriptor)) {
if (output->isConfigurationMatched(mixerAttributes->config, flags)) {
- output->mUsePreferredMixerAttributes = true;
+ output->mPreferredAttrInfo = mixerAttrInfo;
} else {
for (const auto &client: output->getActiveClients()) {
if (client->uid() == uid && client->strategy() == strategy) {
@@ -4737,7 +4814,7 @@
ALOGE("%s, failed to reopen output with preferred mixer attributes", __func__);
continue;
}
- desc->mUsePreferredMixerAttributes = true;
+ desc->mPreferredAttrInfo = mixerAttrInfo;
}
return NO_ERROR;
@@ -4753,8 +4830,7 @@
}
if (activeBitPerfectPreferred) {
for (auto [strategy, info] : it->second) {
- if ((info->getFlags() & AUDIO_OUTPUT_FLAG_BIT_PERFECT) != AUDIO_OUTPUT_FLAG_NONE
- && info->getActiveClientCount() != 0) {
+ if (info->isBitPerfect() && info->getActiveClientCount() != 0) {
return info;
}
}
@@ -5554,7 +5630,7 @@
invalidateStreams(mEngine->getStreamTypesForProductStrategy(ps));
} else {
DeviceVector newDevices = getNewOutputDevices(outputDesc, false /*fromCache*/);
- if (outputDesc->mUsePreferredMixerAttributes && outputDesc->devices() != newDevices) {
+ if (outputDesc->mPreferredAttrInfo != nullptr && outputDesc->devices() != newDevices) {
// If the device is using preferred mixer attributes, the output need to reopen
// with default configuration when the new selected devices are different from
// current routing devices.
@@ -6925,8 +7001,7 @@
closingOutput->stop();
}
closingOutput->close();
- if ((closingOutput->getFlags().output & AUDIO_OUTPUT_FLAG_BIT_PERFECT)
- == AUDIO_OUTPUT_FLAG_BIT_PERFECT) {
+ if (closingOutput->isBitPerfect()) {
for (const auto device : closingOutput->devices()) {
device->setPreferredConfig(nullptr);
}
@@ -6958,6 +7033,10 @@
setMsdOutputPatches();
}
}
+
+ if (closingOutput->mPreferredAttrInfo != nullptr) {
+ closingOutput->mPreferredAttrInfo->resetActiveClient();
+ }
}
void AudioPolicyManager::closeInput(audio_io_handle_t input)
@@ -8815,4 +8894,60 @@
mpClientInterface->invalidateTracks(clients);
}
+void AudioPolicyManager::updateClientsInternalMute(
+ const sp<android::SwAudioOutputDescriptor> &desc) {
+ if (!desc->isBitPerfect() ||
+ !com::android::media::audioserver::
+ fix_concurrent_playback_behavior_with_bit_perfect_client()) {
+ // This is only used for bit perfect output now.
+ return;
+ }
+ sp<TrackClientDescriptor> bitPerfectClient = nullptr;
+ bool bitPerfectClientInternalMute = false;
+ std::vector<media::TrackInternalMuteInfo> clientsInternalMute;
+ for (const sp<TrackClientDescriptor>& client : desc->getActiveClients()) {
+ if ((client->flags() & AUDIO_OUTPUT_FLAG_BIT_PERFECT) != AUDIO_OUTPUT_FLAG_NONE) {
+ bitPerfectClient = client;
+ continue;
+ }
+ bool muted = false;
+ if (client->stream() == AUDIO_STREAM_SYSTEM) {
+ // System sound is muted.
+ muted = true;
+ } else {
+ bitPerfectClientInternalMute = true;
+ }
+ if (client->setInternalMute(muted)) {
+ auto result = legacy2aidl_audio_port_handle_t_int32_t(client->portId());
+ if (!result.ok()) {
+ ALOGE("%s, failed to convert port id(%d) to aidl", __func__, client->portId());
+ continue;
+ }
+ media::TrackInternalMuteInfo info;
+ info.portId = result.value();
+ info.muted = client->getInternalMute();
+ clientsInternalMute.push_back(std::move(info));
+ }
+ }
+ if (bitPerfectClient != nullptr &&
+ bitPerfectClient->setInternalMute(bitPerfectClientInternalMute)) {
+ auto result = legacy2aidl_audio_port_handle_t_int32_t(bitPerfectClient->portId());
+ if (result.ok()) {
+ media::TrackInternalMuteInfo info;
+ info.portId = result.value();
+ info.muted = bitPerfectClient->getInternalMute();
+ clientsInternalMute.push_back(std::move(info));
+ } else {
+ ALOGE("%s, failed to convert port id(%d) of bit perfect client to aidl",
+ __func__, bitPerfectClient->portId());
+ }
+ }
+ if (!clientsInternalMute.empty()) {
+ if (status_t status = mpClientInterface->setTracksInternalMute(clientsInternalMute);
+ status != NO_ERROR) {
+ ALOGE("%s, failed to update tracks internal mute, err=%d", __func__, status);
+ }
+ }
+}
+
} // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 2d015d3..83597d8 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -1386,6 +1386,8 @@
bool checkHapticCompatibilityOnSpatializerOutput(const audio_config_t* config,
audio_session_t sessionId) const;
+
+ void updateClientsInternalMute(const sp<SwAudioOutputDescriptor>& desc);
};
};
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index 6de71a3..f70dc52 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -352,4 +352,13 @@
return af->getAudioMixPort(devicePort, port);
}
+status_t AudioPolicyService::AudioPolicyClient::setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& tracksInternalMute) {
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == nullptr) {
+ return PERMISSION_DENIED;
+ }
+ return af->setTracksInternalMute(tracksInternalMute);
+}
+
} // namespace android
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 5297e47..eb9d081 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -872,6 +872,9 @@
status_t getAudioMixPort(const struct audio_port_v7 *devicePort,
struct audio_port_v7 *port) override;
+ status_t setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& tracksInternalMute) override;
+
private:
AudioPolicyService *mAudioPolicyService;
};
diff --git a/services/audiopolicy/tests/Android.bp b/services/audiopolicy/tests/Android.bp
index fc349ee..dc61115 100644
--- a/services/audiopolicy/tests/Android.bp
+++ b/services/audiopolicy/tests/Android.bp
@@ -40,6 +40,7 @@
static_libs: [
"android.media.audiopolicy-aconfig-cc",
"audioclient-types-aidl-cpp",
+ "com.android.media.audioserver-aconfig-cc",
"libaudiopolicycomponents",
"libflagtest",
"libgmock",
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index 7ef0266..30894c1 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -221,6 +221,15 @@
return NO_ERROR;
}
+ status_t setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& tracksInternalMute) override {
+ for (const auto& trackInternalMute : tracksInternalMute) {
+ mTracksInternalMute[(audio_port_handle_t)trackInternalMute.portId] =
+ trackInternalMute.muted;
+ }
+ return NO_ERROR;
+ }
+
void addSupportedFormat(audio_format_t format) {
mSupportedFormats.insert(format);
}
@@ -229,6 +238,11 @@
mSupportedChannelMasks.insert(channelMask);
}
+ bool getTrackInternalMute(audio_port_handle_t portId) {
+ auto it = mTracksInternalMute.find(portId);
+ return it == mTracksInternalMute.end() ? false : it->second;
+ }
+
private:
audio_module_handle_t mNextModuleHandle = AUDIO_MODULE_HANDLE_NONE + 1;
audio_io_handle_t mNextIoHandle = AUDIO_IO_HANDLE_NONE + 1;
@@ -241,6 +255,7 @@
std::vector<struct audio_port_v7> mDisconnectedDevicePorts;
std::set<audio_format_t> mSupportedFormats;
std::set<audio_channel_mask_t> mSupportedChannelMasks;
+ std::map<audio_port_handle_t, bool> mTracksInternalMute;
};
} // namespace android
diff --git a/services/audiopolicy/tests/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index e55e935..c15adcb 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -110,6 +110,11 @@
struct audio_port_v7 *mixPort __unused) override {
return INVALID_OPERATION;
}
+
+ status_t setTracksInternalMute(
+ const std::vector<media::TrackInternalMuteInfo>& /*tracksInternalMute*/) override {
+ return INVALID_OPERATION;
+ }
};
} // namespace android
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 5dc039c..4f9e98e 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -29,6 +29,7 @@
#include <android-base/properties.h>
#include <android/content/AttributionSourceState.h>
#include <android_media_audiopolicy.h>
+#include <com_android_media_audioserver.h>
#include <flag_macros.h>
#include <hardware/audio_effect.h>
#include <media/AudioPolicy.h>
@@ -1155,129 +1156,6 @@
"", "", AUDIO_FORMAT_LDAC));
}
-TEST_F(AudioPolicyManagerTestWithConfigurationFile, BitPerfectPlayback) {
- const audio_format_t bitPerfectFormat = AUDIO_FORMAT_PCM_16_BIT;
- const audio_channel_mask_t bitPerfectChannelMask = AUDIO_CHANNEL_OUT_QUAD;
- const uint32_t bitPerfectSampleRate = 48000;
- mClient->addSupportedFormat(bitPerfectFormat);
- mClient->addSupportedChannelMask(bitPerfectChannelMask);
- ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(AUDIO_DEVICE_OUT_USB_DEVICE,
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- "", "", AUDIO_FORMAT_DEFAULT));
- auto devices = mManager->getAvailableOutputDevices();
- audio_port_handle_t usbPortId = AUDIO_PORT_HANDLE_NONE;
- for (auto device : devices) {
- if (device->type() == AUDIO_DEVICE_OUT_USB_DEVICE) {
- usbPortId = device->getId();
- break;
- }
- }
- EXPECT_NE(AUDIO_PORT_HANDLE_NONE, usbPortId);
-
- const uid_t uid = 1234;
- const uid_t anotherUid = 5678;
- const audio_attributes_t mediaAttr = {
- .content_type = AUDIO_CONTENT_TYPE_MUSIC,
- .usage = AUDIO_USAGE_MEDIA,
- };
-
- std::vector<audio_mixer_attributes_t> mixerAttributes;
- EXPECT_EQ(NO_ERROR, mManager->getSupportedMixerAttributes(usbPortId, mixerAttributes));
- EXPECT_GT(mixerAttributes.size(), 0);
- size_t bitPerfectIndex = 0;
- for (; bitPerfectIndex < mixerAttributes.size(); ++bitPerfectIndex) {
- if (mixerAttributes[bitPerfectIndex].mixer_behavior == AUDIO_MIXER_BEHAVIOR_BIT_PERFECT) {
- break;
- }
- }
- EXPECT_LT(bitPerfectIndex, mixerAttributes.size());
- EXPECT_EQ(bitPerfectFormat, mixerAttributes[bitPerfectIndex].config.format);
- EXPECT_EQ(bitPerfectChannelMask, mixerAttributes[bitPerfectIndex].config.channel_mask);
- EXPECT_EQ(bitPerfectSampleRate, mixerAttributes[bitPerfectIndex].config.sample_rate);
- EXPECT_EQ(NO_ERROR,
- mManager->setPreferredMixerAttributes(
- &mediaAttr, usbPortId, uid, &mixerAttributes[bitPerfectIndex]));
-
- audio_io_handle_t bitPerfectOutput = AUDIO_IO_HANDLE_NONE;
- audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
- audio_port_handle_t bitPerfectPortId = AUDIO_PORT_HANDLE_NONE;
- audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
- bool isBitPerfect;
-
- // When there is no active bit-perfect playback, the output selection will follow default
- // routing strategy.
- getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
- 48000, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr, AUDIO_SESSION_NONE,
- uid, &isBitPerfect);
- EXPECT_FALSE(isBitPerfect);
- EXPECT_NE(AUDIO_IO_HANDLE_NONE, output);
- const auto outputDesc = mManager->getOutputs().valueFor(output);
- EXPECT_NE(nullptr, outputDesc);
- EXPECT_NE(AUDIO_OUTPUT_FLAG_BIT_PERFECT, outputDesc->mFlags & AUDIO_OUTPUT_FLAG_BIT_PERFECT);
-
- // Start bit-perfect playback
- getOutputForAttr(&selectedDeviceId, bitPerfectFormat, bitPerfectChannelMask,
- bitPerfectSampleRate, AUDIO_OUTPUT_FLAG_NONE, &bitPerfectOutput, &bitPerfectPortId,
- mediaAttr, AUDIO_SESSION_NONE, uid, &isBitPerfect);
- status_t status = mManager->startOutput(bitPerfectPortId);
- if (status == DEAD_OBJECT) {
- getOutputForAttr(&selectedDeviceId, bitPerfectFormat, bitPerfectChannelMask,
- bitPerfectSampleRate, AUDIO_OUTPUT_FLAG_NONE, &bitPerfectOutput, &bitPerfectPortId,
- mediaAttr, AUDIO_SESSION_NONE, uid, &isBitPerfect);
- status = mManager->startOutput(bitPerfectPortId);
- }
- EXPECT_EQ(NO_ERROR, status);
- EXPECT_TRUE(isBitPerfect);
- EXPECT_NE(AUDIO_IO_HANDLE_NONE, bitPerfectOutput);
- const auto bitPerfectOutputDesc = mManager->getOutputs().valueFor(bitPerfectOutput);
- EXPECT_NE(nullptr, bitPerfectOutputDesc);
- EXPECT_EQ(AUDIO_OUTPUT_FLAG_BIT_PERFECT,
- bitPerfectOutputDesc->mFlags & AUDIO_OUTPUT_FLAG_BIT_PERFECT);
-
- // If the playback is from preferred mixer attributes owner but the request doesn't match
- // preferred mixer attributes, it will not be bit-perfect.
- getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
- 48000, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr, AUDIO_SESSION_NONE,
- uid, &isBitPerfect);
- EXPECT_FALSE(isBitPerfect);
- EXPECT_EQ(bitPerfectOutput, output);
-
- // When bit-perfect playback is active, all other playback will be routed to bit-perfect output.
- getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
- 48000, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr, AUDIO_SESSION_NONE,
- anotherUid, &isBitPerfect);
- EXPECT_FALSE(isBitPerfect);
- EXPECT_EQ(bitPerfectOutput, output);
-
- const audio_attributes_t dtmfAttr = {
- .content_type = AUDIO_CONTENT_TYPE_UNKNOWN,
- .usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
- };
- audio_io_handle_t dtmfOutput = AUDIO_IO_HANDLE_NONE;
- selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
- portId = AUDIO_PORT_HANDLE_NONE;
- getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
- 48000, AUDIO_OUTPUT_FLAG_NONE, &dtmfOutput, &portId, dtmfAttr,
- AUDIO_SESSION_NONE, anotherUid, &isBitPerfect);
- EXPECT_FALSE(isBitPerfect);
- EXPECT_EQ(bitPerfectOutput, dtmfOutput);
-
- // When configuration matches preferred mixer attributes, which is bit-perfect, but the client
- // is not the owner of preferred mixer attributes, the playback will not be bit-perfect.
- getOutputForAttr(&selectedDeviceId, bitPerfectFormat, bitPerfectChannelMask,
- bitPerfectSampleRate, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr,
- AUDIO_SESSION_NONE, anotherUid, &isBitPerfect);
- EXPECT_FALSE(isBitPerfect);
- EXPECT_EQ(bitPerfectOutput, output);
-
- EXPECT_EQ(NO_ERROR,
- mManager->clearPreferredMixerAttributes(&mediaAttr, usbPortId, uid));
- ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(AUDIO_DEVICE_OUT_USB_DEVICE,
- AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- "", "", AUDIO_FORMAT_LDAC));
-}
-
TEST_F(AudioPolicyManagerTestWithConfigurationFile, PreferExactConfigForInput) {
const audio_channel_mask_t deviceChannelMask = AUDIO_CHANNEL_IN_3POINT1;
mClient->addSupportedFormat(AUDIO_FORMAT_PCM_16_BIT);
@@ -3612,3 +3490,314 @@
// unregister effect should succeed since effect shall have been restore on the client session
ASSERT_EQ(NO_ERROR, mManager->unregisterEffect(effectId));
}
+
+class AudioPolicyManagerTestBitPerfectBase : public AudioPolicyManagerTestWithConfigurationFile {
+protected:
+ void SetUp() override;
+ void TearDown() override;
+
+ void startBitPerfectOutput();
+ void reset();
+ void getBitPerfectOutput(status_t expected);
+
+ const audio_format_t mBitPerfectFormat = AUDIO_FORMAT_PCM_16_BIT;
+ const audio_channel_mask_t mBitPerfectChannelMask = AUDIO_CHANNEL_OUT_STEREO;
+ const uint32_t mBitPerfectSampleRate = 48000;
+ const uid_t mUid = 1234;
+ audio_port_handle_t mUsbPortId = AUDIO_PORT_HANDLE_NONE;
+
+ audio_io_handle_t mBitPerfectOutput = AUDIO_IO_HANDLE_NONE;
+ audio_port_handle_t mSelectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_port_handle_t mBitPerfectPortId = AUDIO_PORT_HANDLE_NONE;
+
+ static constexpr audio_attributes_t sMediaAttr = {
+ .content_type = AUDIO_CONTENT_TYPE_MUSIC,
+ .usage = AUDIO_USAGE_MEDIA,
+ };
+};
+
+void AudioPolicyManagerTestBitPerfectBase::SetUp() {
+ ASSERT_NO_FATAL_FAILURE(AudioPolicyManagerTestWithConfigurationFile::SetUp());
+
+ mClient->addSupportedFormat(mBitPerfectFormat);
+ mClient->addSupportedChannelMask(mBitPerfectChannelMask);
+ ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(AUDIO_DEVICE_OUT_USB_DEVICE,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ "", "", AUDIO_FORMAT_DEFAULT));
+ auto devices = mManager->getAvailableOutputDevices();
+ mUsbPortId = AUDIO_PORT_HANDLE_NONE;
+ for (auto device : devices) {
+ if (device->type() == AUDIO_DEVICE_OUT_USB_DEVICE) {
+ mUsbPortId = device->getId();
+ break;
+ }
+ }
+ EXPECT_NE(AUDIO_PORT_HANDLE_NONE, mUsbPortId);
+
+ std::vector<audio_mixer_attributes_t> mixerAttributes;
+ EXPECT_EQ(NO_ERROR, mManager->getSupportedMixerAttributes(mUsbPortId, mixerAttributes));
+ EXPECT_GT(mixerAttributes.size(), 0);
+ size_t bitPerfectIndex = 0;
+ for (; bitPerfectIndex < mixerAttributes.size(); ++bitPerfectIndex) {
+ if (mixerAttributes[bitPerfectIndex].mixer_behavior == AUDIO_MIXER_BEHAVIOR_BIT_PERFECT) {
+ break;
+ }
+ }
+ EXPECT_LT(bitPerfectIndex, mixerAttributes.size());
+ EXPECT_EQ(mBitPerfectFormat, mixerAttributes[bitPerfectIndex].config.format);
+ EXPECT_EQ(mBitPerfectChannelMask, mixerAttributes[bitPerfectIndex].config.channel_mask);
+ EXPECT_EQ(mBitPerfectSampleRate, mixerAttributes[bitPerfectIndex].config.sample_rate);
+ EXPECT_EQ(NO_ERROR,
+ mManager->setPreferredMixerAttributes(
+ &sMediaAttr, mUsbPortId, mUid, &mixerAttributes[bitPerfectIndex]));
+}
+
+void AudioPolicyManagerTestBitPerfectBase::TearDown() {
+ EXPECT_EQ(NO_ERROR,
+ mManager->clearPreferredMixerAttributes(&sMediaAttr, mUsbPortId, mUid));
+ ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(AUDIO_DEVICE_OUT_USB_DEVICE,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ "", "", AUDIO_FORMAT_LDAC));
+
+ ASSERT_NO_FATAL_FAILURE(AudioPolicyManagerTestWithConfigurationFile::TearDown());
+}
+
+void AudioPolicyManagerTestBitPerfectBase::startBitPerfectOutput() {
+ reset();
+ bool isBitPerfect;
+
+ getOutputForAttr(&mSelectedDeviceId, mBitPerfectFormat, mBitPerfectChannelMask,
+ mBitPerfectSampleRate, AUDIO_OUTPUT_FLAG_NONE, &mBitPerfectOutput,
+ &mBitPerfectPortId, sMediaAttr, AUDIO_SESSION_NONE, mUid, &isBitPerfect);
+ status_t status = mManager->startOutput(mBitPerfectPortId);
+ if (status == DEAD_OBJECT) {
+ getOutputForAttr(&mSelectedDeviceId, mBitPerfectFormat, mBitPerfectChannelMask,
+ mBitPerfectSampleRate, AUDIO_OUTPUT_FLAG_NONE, &mBitPerfectOutput,
+ &mBitPerfectPortId, sMediaAttr, AUDIO_SESSION_NONE, mUid, &isBitPerfect);
+ status = mManager->startOutput(mBitPerfectPortId);
+ }
+ EXPECT_EQ(NO_ERROR, status);
+ EXPECT_TRUE(isBitPerfect);
+ EXPECT_NE(AUDIO_IO_HANDLE_NONE, mBitPerfectOutput);
+ const auto bitPerfectOutputDesc = mManager->getOutputs().valueFor(mBitPerfectOutput);
+ EXPECT_NE(nullptr, bitPerfectOutputDesc);
+ EXPECT_EQ(AUDIO_OUTPUT_FLAG_BIT_PERFECT,
+ bitPerfectOutputDesc->mFlags & AUDIO_OUTPUT_FLAG_BIT_PERFECT);
+};
+
+void AudioPolicyManagerTestBitPerfectBase::reset() {
+ mBitPerfectOutput = AUDIO_IO_HANDLE_NONE;
+ mSelectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ mBitPerfectPortId = AUDIO_PORT_HANDLE_NONE;
+}
+
+void AudioPolicyManagerTestBitPerfectBase::getBitPerfectOutput(status_t expected) {
+ reset();
+ audio_stream_type_t stream = AUDIO_STREAM_DEFAULT;
+ AttributionSourceState attributionSource = createAttributionSourceState(mUid);
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = mBitPerfectSampleRate;
+ config.channel_mask = mBitPerfectChannelMask;
+ config.format = mBitPerfectFormat;
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_BIT_PERFECT;
+ AudioPolicyInterface::output_type_t outputType;
+ bool isSpatialized;
+ bool isBitPerfect;
+ EXPECT_EQ(expected,
+ mManager->getOutputForAttr(&sMediaAttr, &mBitPerfectOutput, AUDIO_SESSION_NONE,
+ &stream, attributionSource, &config, &flags,
+ &mSelectedDeviceId, &mBitPerfectPortId, {}, &outputType,
+ &isSpatialized, &isBitPerfect));
+}
+
+class AudioPolicyManagerTestBitPerfect : public AudioPolicyManagerTestBitPerfectBase {
+};
+
+TEST_F(AudioPolicyManagerTestBitPerfect, UseBitPerfectOutput) {
+ const uid_t anotherUid = 5678;
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+ bool isBitPerfect;
+
+ // When there is no active bit-perfect playback, the output selection will follow default
+ // routing strategy.
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_QUAD,
+ 48000, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, sMediaAttr,
+ AUDIO_SESSION_NONE, mUid, &isBitPerfect);
+ EXPECT_FALSE(isBitPerfect);
+ EXPECT_NE(AUDIO_IO_HANDLE_NONE, output);
+ const auto outputDesc = mManager->getOutputs().valueFor(output);
+ EXPECT_NE(nullptr, outputDesc);
+ EXPECT_NE(AUDIO_OUTPUT_FLAG_BIT_PERFECT, outputDesc->mFlags & AUDIO_OUTPUT_FLAG_BIT_PERFECT);
+
+ // Start bit-perfect playback
+ ASSERT_NO_FATAL_FAILURE(startBitPerfectOutput());
+
+ // If the playback is from preferred mixer attributes owner but the request doesn't match
+ // preferred mixer attributes, it will not be bit-perfect.
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_QUAD,
+ 48000, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, sMediaAttr,
+ AUDIO_SESSION_NONE, mUid, &isBitPerfect);
+ EXPECT_FALSE(isBitPerfect);
+ EXPECT_EQ(mBitPerfectOutput, output);
+
+ // When bit-perfect playback is active, all other playback will be routed to bit-perfect output.
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ 48000, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, sMediaAttr,
+ AUDIO_SESSION_NONE, anotherUid, &isBitPerfect);
+ EXPECT_FALSE(isBitPerfect);
+ EXPECT_EQ(mBitPerfectOutput, output);
+
+ // When bit-pefect playback is active, dtmf will also be routed to bit-perfect output.
+ const audio_attributes_t dtmfAttr = {
+ .content_type = AUDIO_CONTENT_TYPE_UNKNOWN,
+ .usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
+ };
+ audio_io_handle_t dtmfOutput = AUDIO_IO_HANDLE_NONE;
+ selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ portId = AUDIO_PORT_HANDLE_NONE;
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ 48000, AUDIO_OUTPUT_FLAG_NONE, &dtmfOutput, &portId, dtmfAttr,
+ AUDIO_SESSION_NONE, anotherUid, &isBitPerfect);
+ EXPECT_FALSE(isBitPerfect);
+ EXPECT_EQ(mBitPerfectOutput, dtmfOutput);
+
+ // When configuration matches preferred mixer attributes, which is bit-perfect, but the client
+ // is not the owner of preferred mixer attributes, the playback will not be bit-perfect.
+ getOutputForAttr(&selectedDeviceId, mBitPerfectFormat, mBitPerfectChannelMask,
+ mBitPerfectSampleRate, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, sMediaAttr,
+ AUDIO_SESSION_NONE, anotherUid, &isBitPerfect);
+ EXPECT_FALSE(isBitPerfect);
+ EXPECT_EQ(mBitPerfectOutput, output);
+}
+
+TEST_F_WITH_FLAGS(
+ AudioPolicyManagerTestBitPerfect,
+ InternalMuteWhenBitPerfectCLientIsActive,
+ REQUIRES_FLAGS_ENABLED(
+ ACONFIG_FLAG(com::android::media::audioserver,
+ fix_concurrent_playback_behavior_with_bit_perfect_client))
+) {
+ ASSERT_NO_FATAL_FAILURE(startBitPerfectOutput());
+
+ // When bit-perfect playback is active, the system sound will be routed to bit-perfect output.
+ // The system sound will be muted internally in this case. The bit-perfect client will be
+ // played normally.
+ const uint32_t anotherSampleRate = 44100;
+ audio_port_handle_t systemSoundPortId = AUDIO_PORT_HANDLE_NONE;
+ audio_io_handle_t systemSoundOutput = AUDIO_IO_HANDLE_NONE;
+ const audio_attributes_t systemSoundAttr = {
+ .content_type = AUDIO_CONTENT_TYPE_SONIFICATION,
+ .usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION,
+ };
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ bool isBitPerfect;
+ getOutputForAttr(&selectedDeviceId, mBitPerfectFormat, mBitPerfectChannelMask,
+ anotherSampleRate, AUDIO_OUTPUT_FLAG_NONE, &systemSoundOutput,
+ &systemSoundPortId, systemSoundAttr, AUDIO_SESSION_NONE, mUid, &isBitPerfect);
+ EXPECT_FALSE(isBitPerfect);
+ EXPECT_EQ(mBitPerfectOutput, systemSoundOutput);
+ EXPECT_EQ(NO_ERROR, mManager->startOutput(systemSoundPortId));
+ EXPECT_TRUE(mClient->getTrackInternalMute(systemSoundPortId));
+ EXPECT_FALSE(mClient->getTrackInternalMute(mBitPerfectPortId));
+ EXPECT_EQ(NO_ERROR, mManager->stopOutput(systemSoundPortId));
+ EXPECT_FALSE(mClient->getTrackInternalMute(mBitPerfectPortId));
+
+ // When bit-perfect playback is active, the notification will be routed to bit-perfect output.
+ // The notification sound will be played normally while the bit-perfect client will be muted
+ // internally.
+ audio_port_handle_t notificationPortId = AUDIO_PORT_HANDLE_NONE;
+ audio_io_handle_t notificationOutput = AUDIO_IO_HANDLE_NONE;
+ const audio_attributes_t notificationAttr = {
+ .content_type = AUDIO_CONTENT_TYPE_SONIFICATION,
+ .usage = AUDIO_USAGE_NOTIFICATION,
+ };
+ getOutputForAttr(&selectedDeviceId, mBitPerfectFormat, mBitPerfectChannelMask,
+ anotherSampleRate, AUDIO_OUTPUT_FLAG_NONE, ¬ificationOutput,
+ ¬ificationPortId, notificationAttr, AUDIO_SESSION_NONE, mUid,
+ &isBitPerfect);
+ EXPECT_FALSE(isBitPerfect);
+ EXPECT_EQ(mBitPerfectOutput, notificationOutput);
+ EXPECT_EQ(NO_ERROR, mManager->startOutput(notificationPortId));
+ EXPECT_FALSE(mClient->getTrackInternalMute(notificationPortId));
+ EXPECT_TRUE(mClient->getTrackInternalMute(mBitPerfectPortId));
+ EXPECT_EQ(NO_ERROR, mManager->stopOutput(notificationPortId));
+ EXPECT_FALSE(mClient->getTrackInternalMute(mBitPerfectPortId));
+
+ EXPECT_EQ(NO_ERROR, mManager->stopOutput(mBitPerfectPortId));
+}
+
+class AudioPolicyManagerTestBitPerfectPhoneMode : public AudioPolicyManagerTestBitPerfectBase,
+ public testing::WithParamInterface<audio_mode_t> {
+};
+
+TEST_P(AudioPolicyManagerTestBitPerfectPhoneMode, RejectBitPerfectWhenPhoneModeIsNotNormal) {
+ if (!com::android::media::audioserver::
+ fix_concurrent_playback_behavior_with_bit_perfect_client()) {
+ GTEST_SKIP()
+ << "Flag fix_concurrent_playback_behavior_with_bit_perfect_client is not enabled";
+ }
+
+ ASSERT_NO_FATAL_FAILURE(startBitPerfectOutput());
+
+ audio_mode_t mode = GetParam();
+ mManager->setPhoneState(mode);
+ // When the phone mode is not normal, the bit-perfect output will be reopned
+ EXPECT_EQ(nullptr, mManager->getOutputs().valueFor(mBitPerfectOutput));
+
+ // When the phone mode is not normal, the bit-perfect output will be closed.
+ ASSERT_NO_FATAL_FAILURE(getBitPerfectOutput(INVALID_OPERATION));
+
+ mManager->setPhoneState(AUDIO_MODE_NORMAL);
+}
+
+INSTANTIATE_TEST_CASE_P(
+ PhoneMode,
+ AudioPolicyManagerTestBitPerfectPhoneMode,
+ testing::Values(AUDIO_MODE_IN_CALL,
+ AUDIO_MODE_RINGTONE,
+ AUDIO_MODE_IN_COMMUNICATION,
+ AUDIO_MODE_CALL_SCREEN)
+);
+
+class AudioPolicyManagerTestBitPerfectHigherPriorityUseCaseActive :
+ public AudioPolicyManagerTestBitPerfectBase,
+ public testing::WithParamInterface<audio_usage_t> {
+};
+
+TEST_P(AudioPolicyManagerTestBitPerfectHigherPriorityUseCaseActive,
+ RejectBitPerfectWhenHigherPriorityUseCaseIsActive) {
+ if (!com::android::media::audioserver::
+ fix_concurrent_playback_behavior_with_bit_perfect_client()) {
+ GTEST_SKIP()
+ << "Flag fix_concurrent_playback_behavior_with_bit_perfect_client is not enabled";
+ }
+
+ ASSERT_NO_FATAL_FAILURE(startBitPerfectOutput());
+
+ audio_attributes_t attr = {
+ .usage = GetParam(),
+ .content_type = AUDIO_CONTENT_TYPE_UNKNOWN
+ };
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ ASSERT_NO_FATAL_FAILURE(
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ 48000, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, attr));
+ EXPECT_NE(mBitPerfectOutput, output);
+ EXPECT_EQ(NO_ERROR, mManager->startOutput(portId));
+ // When a high priority use case is active, the bit-perfect output will be closed.
+ EXPECT_EQ(nullptr, mManager->getOutputs().valueFor(mBitPerfectOutput));
+
+ // When any higher priority use case is active, the bit-perfect request will be rejected.
+ ASSERT_NO_FATAL_FAILURE(getBitPerfectOutput(INVALID_OPERATION));
+}
+
+INSTANTIATE_TEST_CASE_P(
+ HigherPriorityUseCases,
+ AudioPolicyManagerTestBitPerfectHigherPriorityUseCaseActive,
+ testing::Values(AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE,
+ AUDIO_USAGE_ALARM)
+);
diff --git a/services/camera/libcameraservice/api2/DepthCompositeStream.cpp b/services/camera/libcameraservice/api2/DepthCompositeStream.cpp
index 1bd0b85..8b41d00 100644
--- a/services/camera/libcameraservice/api2/DepthCompositeStream.cpp
+++ b/services/camera/libcameraservice/api2/DepthCompositeStream.cpp
@@ -517,6 +517,15 @@
return false;
}
+bool DepthCompositeStream::isDepthCompositeStreamInfo(const OutputStreamInfo& streamInfo) {
+ if ((streamInfo.dataSpace == static_cast<android_dataspace_t>(HAL_DATASPACE_DYNAMIC_DEPTH)) &&
+ (streamInfo.format == HAL_PIXEL_FORMAT_BLOB)) {
+ return true;
+ }
+
+ return false;
+}
+
static bool setContains(std::unordered_set<int32_t> containerSet, int32_t value) {
return containerSet.find(value) != containerSet.end();
}
diff --git a/services/camera/libcameraservice/api2/DepthCompositeStream.h b/services/camera/libcameraservice/api2/DepthCompositeStream.h
index f797f9c..3f8f6a2 100644
--- a/services/camera/libcameraservice/api2/DepthCompositeStream.h
+++ b/services/camera/libcameraservice/api2/DepthCompositeStream.h
@@ -46,6 +46,7 @@
~DepthCompositeStream() override;
static bool isDepthCompositeStream(const sp<Surface> &surface);
+ static bool isDepthCompositeStreamInfo(const OutputStreamInfo& streamInfo);
// CompositeStream overrides
status_t createInternalStreams(const std::vector<sp<Surface>>& consumers,
diff --git a/services/camera/libcameraservice/api2/HeicCompositeStream.cpp b/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
index 68e9ad4..225d7f5 100644
--- a/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
+++ b/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
@@ -94,6 +94,11 @@
mMainImageSurface.clear();
}
+bool HeicCompositeStream::isHeicCompositeStreamInfo(const OutputStreamInfo& streamInfo) {
+ return ((streamInfo.dataSpace == static_cast<android_dataspace_t>(HAL_DATASPACE_HEIF)) &&
+ (streamInfo.format == HAL_PIXEL_FORMAT_BLOB));
+}
+
bool HeicCompositeStream::isHeicCompositeStream(const sp<Surface> &surface) {
ANativeWindow *anw = surface.get();
status_t err;
diff --git a/services/camera/libcameraservice/api2/HeicCompositeStream.h b/services/camera/libcameraservice/api2/HeicCompositeStream.h
index b539cdd..9cfd78b 100644
--- a/services/camera/libcameraservice/api2/HeicCompositeStream.h
+++ b/services/camera/libcameraservice/api2/HeicCompositeStream.h
@@ -42,6 +42,7 @@
~HeicCompositeStream() override;
static bool isHeicCompositeStream(const sp<Surface> &surface);
+ static bool isHeicCompositeStreamInfo(const OutputStreamInfo& streamInfo);
status_t createInternalStreams(const std::vector<sp<Surface>>& consumers,
bool hasDeferredConsumer, uint32_t width, uint32_t height, int format,
diff --git a/services/camera/libcameraservice/api2/JpegRCompositeStream.cpp b/services/camera/libcameraservice/api2/JpegRCompositeStream.cpp
index a1b9383..dadefcc 100644
--- a/services/camera/libcameraservice/api2/JpegRCompositeStream.cpp
+++ b/services/camera/libcameraservice/api2/JpegRCompositeStream.cpp
@@ -520,6 +520,15 @@
return false;
}
+bool JpegRCompositeStream::isJpegRCompositeStreamInfo(const OutputStreamInfo& streamInfo) {
+ if ((streamInfo.format == HAL_PIXEL_FORMAT_BLOB) &&
+ (streamInfo.dataSpace == static_cast<int>(kJpegRDataSpace))) {
+ return true;
+ }
+
+ return false;
+}
+
void JpegRCompositeStream::deriveDynamicRangeAndDataspace(int64_t dynamicProfile,
int64_t* /*out*/dynamicRange, int64_t* /*out*/dataSpace) {
if ((dynamicRange == nullptr) || (dataSpace == nullptr)) {
@@ -832,8 +841,8 @@
(*compositeOutput)[0].colorSpace =
ANDROID_REQUEST_AVAILABLE_COLOR_SPACE_PROFILES_MAP_UNSPECIFIED;
- if (CameraProviderManager::isConcurrentDynamicRangeCaptureSupported(staticInfo,
- streamInfo.dynamicRangeProfile,
+ if (CameraProviderManager::isConcurrentDynamicRangeCaptureSupported(
+ staticInfo, dynamicRange,
ANDROID_REQUEST_AVAILABLE_DYNAMIC_RANGE_PROFILES_MAP_STANDARD)) {
compositeOutput->push_back({});
(*compositeOutput)[1].width = streamInfo.width;
diff --git a/services/camera/libcameraservice/api2/JpegRCompositeStream.h b/services/camera/libcameraservice/api2/JpegRCompositeStream.h
index 016d57c..6669739 100644
--- a/services/camera/libcameraservice/api2/JpegRCompositeStream.h
+++ b/services/camera/libcameraservice/api2/JpegRCompositeStream.h
@@ -43,6 +43,7 @@
~JpegRCompositeStream() override;
static bool isJpegRCompositeStream(const sp<Surface> &surface);
+ static bool isJpegRCompositeStreamInfo(const OutputStreamInfo& streamInfo);
// CompositeStream overrides
status_t createInternalStreams(const std::vector<sp<Surface>>& consumers,
diff --git a/services/camera/libcameraservice/utils/SessionConfigurationUtils.cpp b/services/camera/libcameraservice/utils/SessionConfigurationUtils.cpp
index a7a2b5e..40ca276 100644
--- a/services/camera/libcameraservice/utils/SessionConfigurationUtils.cpp
+++ b/services/camera/libcameraservice/utils/SessionConfigurationUtils.cpp
@@ -21,6 +21,7 @@
#include "../api2/HeicCompositeStream.h"
#include "aidl/android/hardware/graphics/common/Dataspace.h"
#include "api2/JpegRCompositeStream.h"
+#include "binder/Status.h"
#include "common/CameraDeviceBase.h"
#include "common/HalConversionsTemplated.h"
#include "../CameraService.h"
@@ -679,6 +680,67 @@
stream->useCase = static_cast<StreamUseCases>(streamInfo.streamUseCase);
}
+binder::Status mapStream(const OutputStreamInfo& streamInfo, bool isCompositeJpegRDisabled,
+ const CameraMetadata& deviceInfo, camera_stream_rotation_t rotation,
+ size_t* streamIdx/*out*/, const std::string &physicalId, int32_t groupId,
+ const std::string& logicalCameraId,
+ aidl::android::hardware::camera::device::StreamConfiguration &streamConfiguration /*out*/,
+ bool *earlyExit /*out*/) {
+ bool isDepthCompositeStream =
+ camera3::DepthCompositeStream::isDepthCompositeStreamInfo(streamInfo);
+ bool isHeicCompositeStream =
+ camera3::HeicCompositeStream::isHeicCompositeStreamInfo(streamInfo);
+ bool isJpegRCompositeStream =
+ camera3::JpegRCompositeStream::isJpegRCompositeStreamInfo(streamInfo) &&
+ !isCompositeJpegRDisabled;
+ if (isDepthCompositeStream || isHeicCompositeStream || isJpegRCompositeStream) {
+ // We need to take in to account that composite streams can have
+ // additional internal camera streams.
+ std::vector<OutputStreamInfo> compositeStreams;
+ status_t ret;
+ if (isDepthCompositeStream) {
+ // TODO: Take care of composite streams.
+ ret = camera3::DepthCompositeStream::getCompositeStreamInfo(streamInfo,
+ deviceInfo, &compositeStreams);
+ } else if (isHeicCompositeStream) {
+ ret = camera3::HeicCompositeStream::getCompositeStreamInfo(streamInfo,
+ deviceInfo, &compositeStreams);
+ } else {
+ ret = camera3::JpegRCompositeStream::getCompositeStreamInfo(streamInfo,
+ deviceInfo, &compositeStreams);
+ }
+
+ if (ret != OK) {
+ std::string msg = fmt::sprintf(
+ "Camera %s: Failed adding composite streams: %s (%d)",
+ logicalCameraId.c_str(), strerror(-ret), ret);
+ ALOGE("%s: %s", __FUNCTION__, msg.c_str());
+ return STATUS_ERROR(CameraService::ERROR_ILLEGAL_ARGUMENT, msg.c_str());
+ }
+
+ if (compositeStreams.size() == 0) {
+ // No internal streams means composite stream not
+ // supported.
+ *earlyExit = true;
+ return binder::Status::ok();
+ } else if (compositeStreams.size() > 1) {
+ size_t streamCount = streamConfiguration.streams.size() + compositeStreams.size() - 1;
+ streamConfiguration.streams.resize(streamCount);
+ }
+
+ for (const auto& compositeStream : compositeStreams) {
+ mapStreamInfo(compositeStream, rotation,
+ physicalId, groupId,
+ &streamConfiguration.streams[(*streamIdx)++]);
+ }
+ } else {
+ mapStreamInfo(streamInfo, rotation,
+ physicalId, groupId, &streamConfiguration.streams[(*streamIdx)++]);
+ }
+
+ return binder::Status::ok();
+}
+
binder::Status
convertToHALStreamCombination(
const SessionConfiguration& sessionConfiguration,
@@ -831,8 +893,13 @@
"Deferred surface sensor pixel modes not valid");
}
streamInfo.streamUseCase = streamUseCase;
- mapStreamInfo(streamInfo, camera3::CAMERA_STREAM_ROTATION_0, physicalCameraId, groupId,
- &streamConfiguration.streams[streamIdx++]);
+ auto status = mapStream(streamInfo, isCompositeJpegRDisabled, deviceInfo,
+ camera3::CAMERA_STREAM_ROTATION_0, &streamIdx, physicalCameraId, groupId,
+ logicalCameraId, streamConfiguration, earlyExit);
+ if (*earlyExit || !status.isOk()) {
+ return status;
+ }
+
isStreamInfoValid = true;
if (numBufferProducers == 0) {
@@ -851,57 +918,11 @@
return res;
if (!isStreamInfoValid) {
- bool isDepthCompositeStream =
- camera3::DepthCompositeStream::isDepthCompositeStream(surface);
- bool isHeicCompositeStream =
- camera3::HeicCompositeStream::isHeicCompositeStream(surface);
- bool isJpegRCompositeStream =
- camera3::JpegRCompositeStream::isJpegRCompositeStream(surface) &&
- !isCompositeJpegRDisabled;
- if (isDepthCompositeStream || isHeicCompositeStream || isJpegRCompositeStream) {
- // We need to take in to account that composite streams can have
- // additional internal camera streams.
- std::vector<OutputStreamInfo> compositeStreams;
- if (isDepthCompositeStream) {
- // TODO: Take care of composite streams.
- ret = camera3::DepthCompositeStream::getCompositeStreamInfo(streamInfo,
- deviceInfo, &compositeStreams);
- } else if (isHeicCompositeStream) {
- ret = camera3::HeicCompositeStream::getCompositeStreamInfo(streamInfo,
- deviceInfo, &compositeStreams);
- } else {
- ret = camera3::JpegRCompositeStream::getCompositeStreamInfo(streamInfo,
- deviceInfo, &compositeStreams);
- }
-
- if (ret != OK) {
- std::string msg = fmt::sprintf(
- "Camera %s: Failed adding composite streams: %s (%d)",
- logicalCameraId.c_str(), strerror(-ret), ret);
- ALOGE("%s: %s", __FUNCTION__, msg.c_str());
- return STATUS_ERROR(CameraService::ERROR_ILLEGAL_ARGUMENT, msg.c_str());
- }
-
- if (compositeStreams.size() == 0) {
- // No internal streams means composite stream not
- // supported.
- *earlyExit = true;
- return binder::Status::ok();
- } else if (compositeStreams.size() > 1) {
- streamCount += compositeStreams.size() - 1;
- streamConfiguration.streams.resize(streamCount);
- }
-
- for (const auto& compositeStream : compositeStreams) {
- mapStreamInfo(compositeStream,
- static_cast<camera_stream_rotation_t> (it.getRotation()),
- physicalCameraId, groupId,
- &streamConfiguration.streams[streamIdx++]);
- }
- } else {
- mapStreamInfo(streamInfo,
- static_cast<camera_stream_rotation_t> (it.getRotation()),
- physicalCameraId, groupId, &streamConfiguration.streams[streamIdx++]);
+ auto status = mapStream(streamInfo, isCompositeJpegRDisabled, deviceInfo,
+ static_cast<camera_stream_rotation_t> (it.getRotation()), &streamIdx,
+ physicalCameraId, groupId, logicalCameraId, streamConfiguration, earlyExit);
+ if (*earlyExit || !status.isOk()) {
+ return status;
}
isStreamInfoValid = true;
}
diff --git a/services/mediametrics/include/mediametricsservice/AudioTypes.h b/services/mediametrics/include/mediametricsservice/AudioTypes.h
index b5fe28b..59654bf 100644
--- a/services/mediametrics/include/mediametricsservice/AudioTypes.h
+++ b/services/mediametrics/include/mediametricsservice/AudioTypes.h
@@ -18,6 +18,7 @@
#include <string>
#include <unordered_map>
+#include <vector>
namespace android::mediametrics::types {
diff --git a/services/mediaresourcemanager/fuzzer/Android.bp b/services/mediaresourcemanager/fuzzer/Android.bp
index 5bac062..3f04f69 100644
--- a/services/mediaresourcemanager/fuzzer/Android.bp
+++ b/services/mediaresourcemanager/fuzzer/Android.bp
@@ -47,7 +47,7 @@
],
fuzz_config: {
cc: [
- "android-media-fuzzing-reports@google.com",
+ "girishshetty@google.com",
],
componentid: 155276,
hotlists: [